search for: zx81

I'm no longer on the DEV mailing list, but: # svn checkout http://svn.digium.com/svn/asterisk/branches/1.4 asterisk svn: URL 'http://svn.digium.com/svn/asterisk/branches/1.4' doesn't exist http://svn.digium.com/svn/asterisk/branches/ -- /Nick -------------- next part -------------- An HTML attachment was scrubbed... URL:

...a significant bug fix update for the 1.4 release series. The following list of bugs were resolved with the participation of the community, and this release would not have been possible without your help! * Fixed datastore handling crashes in app_queue.c. - Closes issues 14086. Reported by ZX81. Tested by: ZX81, festr. * Update autosupport script to supply info for both Zaptel and DAHDI in 1.4 and be sure to run dahdi_test in 1.6.x and trunk instead of zttest. - Closes issue 14132. Reported by and patch contributed by: dsedivec * Don't overflow when paging more tha...

I have been using Asterisk 1.0-RC2 successfully with a channelized T1 circuit for quite a while now but after upgrading to 1.0.0 callerid no longer works properly. Debug output from a channel shows what actually is received through DTMF from the carrier: << [ TYPE: DTMF (1) SUBCLASS: * (42) ] [Zap/9-1] << [ TYPE: DTMF (1) SUBCLASS: 2 (50) ] [Zap/9-1] << [ TYPE: DTMF (1)

Thanks Bruce, and elaborately as ever. Again, I'm surprised about your very thorough Z80 knowledge when you said you only did little on the ZX81 in the eighties :D OK, understood. I was first thinking about doing something like this for small frames: 1. push bc # 1 byte; 11 cycles - part of call frame-cleanup: save scratch register +-----begin call-related 2. ld <rr>,stack-param 3. push <rr> ... more code to...

...chael Stellmann > <Michael.Stellmann at gmx.net <mailto:Michael.Stellmann at gmx.net>> wrote: > > Thanks Bruce, > > and elaborately as ever. Again, I'm surprised about your very > thorough Z80 knowledge when you said you only did little on the > ZX81 in the eighties :D > > OK, understood. I was first thinking about doing something like > this for small frames: > > 1. push bc # 1 byte; 11 cycles - part of call frame-cleanup: save > scratch register > > +-----begin call-related > 2. ld <...

...too few arguments to RetryDial. - Closes issue #14852. Reported, and patched by junky. * Fix RFC2833 issues with DTMF getting duplicated and with duration wrapping over. - Closes issue #14815 and #14460. Reported by geoff2010 and moliveras. Patched by dimas. Tested by geoff2010, file, dimas, ZX81, moliveras. Thank you for your continued support of Asterisk!

...l issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this release: * Fix Deadlock with attended transfer of SIP call (Closes issue #18837. Reported, patched by alecdavis. Tested by Irontec, ZX81, cmaj) * Fixes thread blocking issue in the sip TCP/TLS implementation. (Closes issue #18497. Reported by vois. Patched by dvossel. Tested by vois, rossbeer, kowalma, Freddi_Fonet) * Be more tolerant of what URI we accept for call completion PUBLISH requests. (Closes issue #18946. Reporte...

Hi All, http://www.voip-info.org/wiki/index.php?page=Asterisk+func+cdr Under example: exten => s,2,Set(CDR(MyFavoriteBand)=Foo Fighters) exten => s,3,Set(CDR(MyFavoriteSong)=Hero) and under description: -userfield: The channel's user specified field. ""-any custom value that you wish to store."" My question is how do you setup more custom fields in the cdr and be

...too few arguments to RetryDial. - Closes issue #14852. Reported, and patched by junky. * Fix RFC2833 issues with DTMF getting duplicated and with duration wrapping over. - Closes issue #14815 and #14460. Reported by geoff2010 and moliveras. Patched by dimas. Tested by geoff2010, file, dimas, ZX81, moliveras. Thank you for your continued support of Asterisk!

...l issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this release: * Fix Deadlock with attended transfer of SIP call (Closes issue #18837. Reported, patched by alecdavis. Tested by Irontec, ZX81, cmaj) * Fixes thread blocking issue in the sip TCP/TLS implementation. (Closes issue #18497. Reported by vois. Patched by dvossel. Tested by vois, rossbeer, kowalma, Freddi_Fonet) * Be more tolerant of what URI we accept for call completion PUBLISH requests. (Closes issue #18946. Reporte...

...(Patched by twilson) * Fix issues with verbose messages not being output to the console. (Closes issue #18580. Reported by pabelanger. Patched by qwell) * Fix Deadlock with attended transfer of SIP call (Closes issue #18837. Reported, patched by alecdavis. Tested by alecdavid, Irontec, ZX81, cmaj) Includes changes per AST-2011-005 and AST-2011-006 For a full list of changes in this release candidate, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.4 Information about the security releases are available at: http://downloads.asterisk.org/...

...(Patched by twilson) * Fix issues with verbose messages not being output to the console. (Closes issue #18580. Reported by pabelanger. Patched by qwell) * Fix Deadlock with attended transfer of SIP call (Closes issue #18837. Reported, patched by alecdavis. Tested by alecdavid, Irontec, ZX81, cmaj) Includes changes per AST-2011-005 and AST-2011-006 For a full list of changes in this release candidate, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.4 Information about the security releases are available at: http://downloads.asterisk.org/...

A customer of mine has 3 TDM400P cards in a box running asterisk. On each card he has four FXO modules. I have set up the dialplan to dial via group 1 for an outgoing call. Channels 1-12 are in group 1. If he plugs a telephone cable into socket 2 or 3 etc, but not 1, when he dials out, it still tries to make the call via socket 1. Straight away the console says that it has dialed the

As discussed previously: https://www.redhat.com/archives/libguestfs/2010-July/msg00003.html https://www.redhat.com/archives/libguestfs/2010-July/msg00024.html To do: (1) Implement progress notifications for many more daemon operations. (2) OCaml bindings to the callback. (3) Perl bindings to the callback. Rich. -- Richard Jones, Virtualization Group, Red Hat

For a Z80 backend, "eliminateCallFramePseudoInstr()" shall adjust the stack pointer in three possible ways, e.g. after a function call, depending on the amount (= adjustment size) *and some other rules*: 1. via one or more target "pop <reg>" instructions (SP increments +2 per instruction), using an unused reg (disregarding the contents after the operation), followed

Hi..... I am new to this software, and I want to implement a client (SIP or IAX) with PHP or at least to pass the main functions (connection,call, transfer, hangup, call id etc) to a CRM. Does anyone know if I could achive a project like that with AGI ? Any example using AGI with PHP ? Do I have all the functionality with AGI ? What about call id ? What is depend on ? (As I know * does not