similar to: Difference between Application Set and Function SET?

Is this patch correct? The "&&" doesn't make logical sense to me. I think it should be "||" and making this change fixes the problem I have with SIP phones in MeetMe conferences. If it's correct, is there someplace more formal that I should submit it to? *** app_meetme.c.old 2009-10-11 17:56:44.000000000 -0400 --- app_meetme.c 2009-10-17

I'm having a wierd problem. Somewhere around 1-2% of the time, the first DTMF digit dialed gets dropped. This is occurring during a SpeechBackground application call. If the caller reenters the digits when given a second chance, all is OK. Any suggestions how to debug this intermittent problem?

It appears that there are no transcoders from g723 to anything else in Asterisk 10.7.1. Does anybody know how to fix that?

I'm confused exactly what's supported with LDAP and Asterisk. What I want to do is to have SIP peer information read directly (in realtime) from LDAP. Can this be done? If so, with what Asterisk versions?

> CALLERID is a read only variable. That's not correct. I set it all over the place in my dialplan.

Related to http://lists.digium.com/pipermail/asterisk-users/2016-November/290384.html, at the moment I'm passing one variable via DIAL. Now I'd like to pass a whole bunch, and my idea was to rather than having a great string of b(synctest3b^setVar^1(something)^2(more things)^3(etc)) and then get them with ARG1..ARGn etc, I could bundle the whole lot into a HASH and then unbundle them at

Hi, when I'm trying to call the following extension exten => 6002,1,Verbose(1|Extension 6002) exten => 6002,n,Dial(Agent/6002) exten => 6002,n,Hangup() the call is terminated and I get the following warning from asterisk: app_dial.c:1106 dial_exec_full: Unable to create channel of type 'Agent' (cause 17 - User busy) When calling the agent with Dial(SIP/6002) no problem

I am writing a dialplan context under asterisk 11.21.0 to handle SIP message routing between registered SIP peers using chan_sip. I am having trouble with double-quotes when the source peer uses a display name, which appears in quotes before the SIP URI. I want to extract the SIP URI from MESSAGE(from) in order to (conditionally) route a failure message back to the source peer. My test dialplan

On Thu, Dec 15, 2016 at 1:30 PM Teresa Johnson <tejohnson at google.com> wrote: On Thu, Dec 15, 2016 at 11:38 AM, David Blaikie <dblaikie at gmail.com> wrote: On Thu, Dec 15, 2016 at 11:26 AM Teresa Johnson <tejohnson at google.com> wrote: Trying to wrap my brain around this, so a few questions below. =) Sure thing - sorry, did assume a bit too much arcane context here.

On Thu, Dec 15, 2016 at 11:38 AM, David Blaikie <dblaikie at gmail.com> wrote: > > > On Thu, Dec 15, 2016 at 11:26 AM Teresa Johnson <tejohnson at google.com> > wrote: > >> Trying to wrap my brain around this, so a few questions below. =) >> > > Sure thing - sorry, did assume a bit too much arcane context here. > > >> >> On Thu, Dec

It's kind of low for me. How does one control that volume?

On Thu, Dec 15, 2016 at 2:08 PM, David Blaikie <dblaikie at gmail.com> wrote: > On Thu, Dec 15, 2016 at 1:30 PM Teresa Johnson <tejohnson at google.com> > wrote: > > On Thu, Dec 15, 2016 at 11:38 AM, David Blaikie <dblaikie at gmail.com> > wrote: > > > > On Thu, Dec 15, 2016 at 11:26 AM Teresa Johnson <tejohnson at google.com> > wrote: >

Which speed recognition products will also recognize DTMF? In other words, I want to say "Please speak or dial the conference number". Does Vestec allow that? LumenVox? Any other way?

On Thu, Dec 15, 2016 at 4:17 PM Teresa Johnson <tejohnson at google.com> wrote: > On Thu, Dec 15, 2016 at 2:08 PM, David Blaikie <dblaikie at gmail.com> wrote: > > On Thu, Dec 15, 2016 at 1:30 PM Teresa Johnson <tejohnson at google.com> > wrote: > > On Thu, Dec 15, 2016 at 11:38 AM, David Blaikie <dblaikie at gmail.com> > wrote: > > > > On

Hi I was wondering whether I can extract the function signature from the dwarf debugging information or symbol table. I mean the argument list and the return variables. I know that in dwarf debugging information, a subprogram represents a function. However, I could not find the information that is related to the function signature. Any ideas? Many Thanks Regards Muhui -------------- next part

On Thu, Dec 15, 2016 at 4:20 PM, David Blaikie <dblaikie at gmail.com> wrote: > > > On Thu, Dec 15, 2016 at 4:17 PM Teresa Johnson <tejohnson at google.com> > wrote: > >> On Thu, Dec 15, 2016 at 2:08 PM, David Blaikie <dblaikie at gmail.com> >> wrote: >> >> On Thu, Dec 15, 2016 at 1:30 PM Teresa Johnson <tejohnson at google.com>

I'm confused. What does Asterisk do when it gets a 302 with a new number to forward to? Is there anything I have to do in the dialplan to make this work? I can't find any clear documentation on this issue.

configure.ac | 5 - libswfdec/Makefile.am | 2 libswfdec/swfdec_as_context.c | 9 +- libswfdec/swfdec_as_debugger.h | 8 ++ libswfdec/swfdec_as_object.c | 5 + libswfdec/swfdec_movie.c | 15 --- libswfdec/swfdec_script.c | 1 libswfdec/swfdec_types.h | 1 vivified/core/vivi_debugger.c | 34

Hi, I am working on an issue where a subprogram created here is having abstract_origin pointing to nowhere. DIE *DwarfDebug::updateSubprogramScopeDIE(CompileUnit *SPCU, const MDNode *SPNode) { DIE *SPDie = SPCU->getDIE(SPNode); assert(SPDie && "Unable to find subprogram DIE!"); DISubprogram SP(SPNode); // If we're

That happens because we create the subprogram below as a context to the “DW_TAG_typedef” that was created as a type to “DW_TAG_pointer_type” that was added to the retained type list because of the explicit cast to (T*). This is the code that creates DW_TAG_subprogram: DIE *DwarfUnit::getOrCreateSubprogramDIE(const DISubprogram *SP, bool Minimal) { ... // DW_TAG_inlined_subroutine may refer