search for: guse

...e-channel gets created. Working old style would be (extensions.conf) [default] exten => _X,1,dial(SIP/1,,M(mymacro^parameter)) [macro-mymacro] exten => s,1,verbose(${ARG1}) How to implement the same functionality using pbx_lua? Details: Asterisk 11.7 on Ubuntu 14.04 Kind regards Dennis Guse Quality and Usability Lab Telekom Innovation Laboratories TU Berlin Ernst-Reuter-Platz 7 D-10587 Berlin, Germany Tel: +49 30 8353 58874 Fax: +49 30 8353 58409 E-mail: dennis.guse at telekom.de Web: www.qu.tlabs.tu-berlin.de -------------- next part -------------- An HTML attachment was scrubbed......

...orked quite well, but I now checked the system again and it is only obeying the first register statement. "sip show registry" only reports the first entry and if I reorder them, this effect stays the same. Did something changed recently in the parsing code for sip.conf or so? --- Dennis Guse -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150317/3a5c6395/attachment.html>

...tion unexpectedly closed (0 bytes received so far) [receiver] rsync error: error in rsync protocol data stream (code 12) at io.c(434) I have tried different parameter, but also with a normal telnet got the same message. Have you any advices for my little odd problem? Thanks for your time. Dennis Guse --FHTW-University of Applied Science Berlin--

Any one know is there any package or function to generate bivariate exponential distribution? I gusee there should be three parameters, two rate parameters and one correlation parameter. I just did not find any function available on R. Any suggestion is appreciated. -- View this message in context: http://r.789695.n4.nabble.com/How-to-generate-bivariate-exponential-distribution-tp3596944p359694...

Hi, I use Asterisk to create the dial tone (indications.conf), which works quite well. However the generated signal is quite loud at the client side (in comparison to the following speech ). Is there an option to modify the volume? --- Dennis Guse -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140422/f036117b/attachment.html>

...\ --with-ssldir=/etc/certs \ --with-shadow \ --with-pam \ --with-ldap \ --enable-header-install Thanks, Laurent -- / Leader de Projet & Communaut? | I'm working, but not speaking for \ G11N http://fr.opensolaris.org | Bull Services http://www.bull.com / FOSUG http://guses.org |

...ota2 = fs:Others:mount=/export/home ---- quota = fs:Others:mount=/export/home Any idea what I'm doing wrong? TIA, Laurent -- / Leader de Projet & Communaut? | I'm working, but not speaking for \ G11N http://fr.opensolaris.org | Bull Services http://www.bull.com / FOSUG http://guses.org |

...more details if needed: http://www.blastwave.org/mantis/view_bug_page.php?f_id=0002760 Thanks in advance for any help, Laurent -- / Leader de Projet & Communaut? | I'm working, but not speaking for \ G11N http://fr.opensolaris.org | Bull Services http://www.bull.com / FOSUG http://guses.org |

Hello, I am studing Xen hypercall now. I found that hypercall is invoked via hypercall_page, which is only filled with (in no-hypervisor-kernel-mode ): mov $i, %eax int $0x82 ret Why not invoked the hypercall directly by "int $0x82" ? What''s the advantage of using hypercall_page? Thanks, Wu _______________________________________________ Xen-devel mailing list

..."Others", and the value are 6490 out of 10000. Any idea why this doesn't display the value of /var/mail? Laurent -- / Leader de Projet & Communaut? | I'm working, but not speaking for \ G11N http://fr.opensolaris.org | Bull Services http://www.bull.com / FOSUG http://guses.org |

...9;s dead and doesn't seem interesting. > Also consider the case where the caller of foo() (which does explicitly use E) is in a dylib we haven't loaded yet, and thus haven't looked at the debug info for. I want to set that same breakpoint. -flimit-debug-info=0. The entire idea of -gused is that you're building (almost) all of your app with debug info. It's the default because this matches the common scenario. In short, I still don't understand why we'd want to emit all the enumerators just because an enum type is used. :) -Chris

...: yes passdb: driver: pam args: blocking=yes setcred=yes dovecot userdb: driver: passwd args: blocking=yes -- / Leader de Projet & Communaut? | I'm working, but not speaking for \ G11N http://fr.opensolaris.org | Bull Services http://www.bull.com / FOSUG http://guses.org |

On Nov 9, 2011, at 12:52 PM, Chris Lattner wrote: > > On Nov 9, 2011, at 10:49 AM, Jim Grosbach wrote: >>> >>> It's not good, but people do it. Also constructing enums via & and | etc. It'd be nice to be able to get the name of whatever it is that the code generator actually produced :) >>> >> >> Agreed. LLVM itself does this sort of

...ror] Raw backtrace: 0x24d90 -> 0x2851c -> 0x286ac -> 0x28728 -> 0x28de0 -> 0x2877c -> 0x1f56c -> 0x15c40 -- / Leader de Projet & Communaut? | I'm working, but not speaking for \ G11N http://fr.opensolaris.org | Bull Services http://www.bull.com / FOSUG http://guses.org |

Dennis Guse wrote: > Hey, > > I am running default Asterisk 11.16.0 on a FreeBSD-Machine. > I need to register to several other SIP-Services (actually 3): > > short sip.conf > > register => XX at a > register => XX at b > register => XX at c > > If I remember correc...

Hi Guys, Does anyone know what this error means and how to fix it? [Jul 3 11:57:27] WARNING[17040] pbx.c: Don't know what to do with 'SIP/ -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140703/dfa17d19/attachment.html>

...r peer dialed (Reported by Igor Goncharovsky) * ASTERISK-24087 - [patch]chan_sip: sip_subscribe_mwi_destroy should not call sip_destroy (Reported by Corey Farrell) * ASTERISK-23818 - PBX_Lua: after asterisk startup module is loaded, but dialplan not available (Reported by Dennis Guse) * ASTERISK-18345 - [patch] sips connection dropped by asterisk with a large INVITE (Reported by Stephane Chazelas) * ASTERISK-23508 - Memory Corruption in __ast_string_field_ptr_build_va (Reported by Arnd Schmitter) Improvements made in this release: --------------------------------...

...r peer dialed (Reported by Igor Goncharovsky) * ASTERISK-24087 - [patch]chan_sip: sip_subscribe_mwi_destroy should not call sip_destroy (Reported by Corey Farrell) * ASTERISK-23818 - PBX_Lua: after asterisk startup module is loaded, but dialplan not available (Reported by Dennis Guse) * ASTERISK-18345 - [patch] sips connection dropped by asterisk with a large INVITE (Reported by Stephane Chazelas) * ASTERISK-23508 - Memory Corruption in __ast_string_field_ptr_build_va (Reported by Arnd Schmitter) Improvements made in this release: --------------------------------...

...r peer dialed (Reported by Igor Goncharovsky) * ASTERISK-24087 - [patch]chan_sip: sip_subscribe_mwi_destroy should not call sip_destroy (Reported by Corey Farrell) * ASTERISK-23818 - PBX_Lua: after asterisk startup module is loaded, but dialplan not available (Reported by Dennis Guse) * ASTERISK-18345 - [patch] sips connection dropped by asterisk with a large INVITE (Reported by Stephane Chazelas) * ASTERISK-23508 - Memory Corruption in __ast_string_field_ptr_build_va (Reported by Arnd Schmitter) Improvements made in this release: --------------------------------...

...r peer dialed (Reported by Igor Goncharovsky) * ASTERISK-24087 - [patch]chan_sip: sip_subscribe_mwi_destroy should not call sip_destroy (Reported by Corey Farrell) * ASTERISK-23818 - PBX_Lua: after asterisk startup module is loaded, but dialplan not available (Reported by Dennis Guse) * ASTERISK-18345 - [patch] sips connection dropped by asterisk with a large INVITE (Reported by Stephane Chazelas) * ASTERISK-23508 - Memory Corruption in __ast_string_field_ptr_build_va (Reported by Arnd Schmitter) Improvements made in this release: --------------------------------...