search for: michelson

...le without your participation. Thank you! The following are the issues resolved in this release: New Features made in this release: ----------------------------------- * ASTERISK-25377 - res_pjsip: Change default "From user" from UUID to something more palatable (Reported by Mark Michelson) * ASTERISK-25252 - ARI: Add the ability to manipulate log channels (Reported by Matt Jordan) Bugs fixed in this release: ----------------------------------- * ASTERISK-25449 - main/sched: Regression introduced by 5c713fdf18f causes erroneous duplicate RTCP messages; other pote...

Howdy, Was there ever a fix for this? I have Trix 2.6 running asterisk 1.4 and have to set an extension with call-limit=1. However that user can no longer do attended transfers from Linkys 962 ip phone. Is there anyway around this? Cheers, Taff..

Hi all, I have upgraded my Asterisk box from 1.2.x to 1.4.x version: it seems that sometimes some phones become paused and cannot receive calls anymore. I tried to set autopause = no in every section of my queues.conf but nothing changes.... Anybody knows why a phone becomes paused? Is it an Asterisk 1.4 bug or there is a particular reason for this behaviour? Thank you. Giorgio.

Hi All, Asterisk 1.4.12 on CentOS 5 Sorry for a question that I'm guessing is obvious to most of you. I'm trying to revamp my dialplan. When I first created it, I had something like: exten => s,1,Set(CALLERID(name)=${DB(cidname/${CALLERID(num)})}) exten => s,2,Dial(${rgMain},${RINGTIME},t) exten => s,3,VoiceMail(main at default) exten => s,103,VoiceMail(main at default)

Ok Mark Michelson. Thank you very much! You answer tells me that I was in the wrong path trying to access information from SIP 183 message. I need to find a way to let the callee pass information/data to the caller, even before accepting the call. That is, send data during the ringing time. And in my case, there w...

...ERISK-25480 - [patch]Add field PauseReason on QueueMemberStatus (Reported by Rodrigo Ramirez Norambuena) * ASTERISK-25419 - Dialplan Application for Integration of StatsD (Reported by Ashley Sanders) * ASTERISK-25549 - Confbridge: Add participant timeout option (Reported by Mark Michelson) * ASTERISK-24922 - ARI: Add the ability to intercept hold and raise an event (Reported by Matt Jordan) * ASTERISK-25377 - res_pjsip: Change default "From user" from UUID to something more palatable (Reported by Mark Michelson) * ASTERISK-25252 - ARI: Add the ability to man...

...--------------- * ASTERISK-25904 - PJSIP: add contact.updated event (Reported by Alexei Gradinari) * ASTERISK-26058 - [Patch] Add uptime and last reloaded to FullyBooted AMI event (Reported by Niklas Larsson) * ASTERISK-25925 - Allow Early Bridges on ARI Dials (Reported by Mark Michelson) * ASTERISK-26068 - Multicast RTP Options (Reported by Mark Michelson) * ASTERISK-26042 - ARI: Allow downloading of the media associated with a stored recording (Reported by Matt Jordan) * ASTERISK-25425 - logger: Add JSON structured logging (Reported by Matt Jordan) * ASTERIS...

...oogle groups and in various forums, to no avail. My webserver does send out emails to a few thousand registered users (if they opt it for the email) every day. Thank you for your time! I hope I sent this to the right list. This looked like the right one. Sorry in advance if I made a mistake. Michelson

...s accepted for 'confbridge kick all' (Reported by Dorian Logan) * ASTERISK-19465 - P-Asserted-Identity Privacy (Reported by Krzysztof Chmielewski) * ASTERISK-23573 - Crash when transferring unbridged call - in bridge_app_subscribed at stasis/app.c (Reported by Mark Michelson) * ASTERISK-23639 - PJSIP Realtime: Alembic migration needed in order to widen some string columns (Reported by Mark Michelson) * ASTERISK-23560 - [ARI] MOH doesn't indicate progress (Reported by Jan Svoboda) * ASTERISK-23605 - res_http_websocket: Race condition in shutting...

...s accepted for 'confbridge kick all' (Reported by Dorian Logan) * ASTERISK-19465 - P-Asserted-Identity Privacy (Reported by Krzysztof Chmielewski) * ASTERISK-23573 - Crash when transferring unbridged call - in bridge_app_subscribed at stasis/app.c (Reported by Mark Michelson) * ASTERISK-23639 - PJSIP Realtime: Alembic migration needed in order to widen some string columns (Reported by Mark Michelson) * ASTERISK-23560 - [ARI] MOH doesn't indicate progress (Reported by Jan Svoboda) * ASTERISK-23605 - res_http_websocket: Race condition in shutting...

Hi, I am sorry to say, I do not know where the 'morley.dat' is that is mentioned as the sample data for the sample session on page 80 of Appendix A in the Introduction to R book. If you could get back to me soon about that would be great! Then I can prepare my next question, steve __________________________________________________ Do You Yahoo!? Send online invitations with Yahoo!

...| |----------------------+-------------------------------------------------| | Last Updated On | December 9, 2008 | |----------------------+-------------------------------------------------| | Advisory Contact | Mark Michelson <mmichelson AT digium DOT com> | |----------------------+-------------------------------------------------| | CVE Name | | +------------------------------------------------------------------------+ +-------------------...

Hi, Is it possible to use realtime for Music On Hold? Is it also possible to store the music/audio files on the database, same way a voicemail can be stored on the database? Thank You Regards, Nhadie

I just installed *-1.4 and when I enter mail extension it keep asking me for Mailbox # I have in sip.conf under my extension mailbox=11 type=friend *-1.2 was jumping straight to messages. What did change? -- #Joseph

...| |----------------------+-------------------------------------------------| | Last Updated On | December 9, 2008 | |----------------------+-------------------------------------------------| | Advisory Contact | Mark Michelson <mmichelson AT digium DOT com> | |----------------------+-------------------------------------------------| | CVE Name | | +------------------------------------------------------------------------+ +-------------------...

Hi list, can anyone tell me how to reset/delete all modifications (personal greeting message, personal name, ...) I made in my voicemail? I just want to get the default automatic computer messages back. thank you! greets felix

...sessions Severity Moderate Exploits Known No Reported On 28 May, 2014 Reported By Mark Michelson Posted On June 12, 2014 Last Updated On June 12, 2014 Advisory Contact Mark Michelson <mmichelson AT digium DOT com> CVE Nam...

...sessions Severity Major Exploits Known No Reported On 30 July, 2014 Reported By Mark Michelson Posted On 18 September, 2014 Last Updated On September 18, 2014 Advisory Contact Mark Michelson <mmichelson AT digium DOT com> CVE Nam...

...sessions Severity Moderate Exploits Known No Reported On 28 May, 2014 Reported By Mark Michelson Posted On June 12, 2014 Last Updated On June 12, 2014 Advisory Contact Mark Michelson <mmichelson AT digium DOT com> CVE Nam...

...sessions Severity Major Exploits Known No Reported On 30 July, 2014 Reported By Mark Michelson Posted On 18 September, 2014 Last Updated On September 18, 2014 Advisory Contact Mark Michelson <mmichelson AT digium DOT com> CVE Nam...