Asterisk Internet Phone System
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- asterisk release 21.1.0,
Asterisk Development Team
- asterisk release 20.6.0,
Asterisk Development Team
- asterisk release 18.21.0,
Asterisk Development Team
- Mailing List Shutdown Reminder,
Joshua C. Colp
- aeap wss connection,
marek
- SIP_HEADER GET_TRANSFERRER_DATA chan_pjsip,
marek
- chan_iax2.c:4739 __auto_congest: Auto-congesting call due to slow response,
thelma
- asterisk release 21.0.2,
Asterisk Development Team
- asterisk release 20.5.2,
Asterisk Development Team
- asterisk release 18.20.2,
Asterisk Development Team
- asterisk release certified-18.9-cert7,
Asterisk Development Team
- CORRECTED asterisk release certified-18.9-cert6,
Asterisk Development Team
- asterisk release certified-18.9-cert6,
Asterisk Development Team
- asterisk release 21.0.1,
Asterisk Development Team
- asterisk release 20.5.1,
Asterisk Development Team
- asterisk release 18.20.1,
Asterisk Development Team
- retry loop in ansible ?,
Axel Rau
- Mailing List Future,
Joshua C. Colp
Asterisk 13 / chan_sip / registration after reject,
Benoit Panizzon
Finding old patches,
Dovid Bender
Recommended sip providers,
Tahir Almas Dhesi
Enterqueue event not generated when cfu internal,
Jon Bonilla (Manwe)
help with crash,
Federico
[Maybe OT]: SIP Provider,
Luca Bertoncello
Local calls not possible when Internet connection down,
Marek Greško
Asterisk and Teams integration?,
Carlos Chavez
asterisk release 21.0.0,
Asterisk Development Team
asterisk release 20.5.0,
Asterisk Development Team
asterisk release 18.20.0,
Asterisk Development Team
Deleting voicemail by program,
Mike Diehl
CDR gets lost,
Federico
Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa,
Jerry Geis
Asterisk 16.23.0 strange issue where Answer request succeeds and able to perform actions but Asterisk never sent 200 OK to answer call,
Dan Cropp
asterisk 18.18.0 and chan_console,
Jerry Geis
AstriCon 2024: February 15th, 2024 - Fort Lauderdale, Florida,
Joshua C. Colp
Saving "admins" from themselves,
Dovid Bender
Question on the RTP packet header,
Dan Cropp
ICE Candidate collision on dualstack hosts?,
Benoît Panizzon
Quick patch for updated NL-ips,
Dirk-Willem van Gulik
Some links on new docs asterisk org not working,
Dan Cropp
60+ devices in confbridge and dropping audio,
Jerry Geis
PJSIP Losing knowledge of external_media_address,
Mark Murawski
Question about Sip Trunks who support Stir Shaken,
Federico
Alternative to Local channel,
Federico
libpri release 1.6.1,
Asterisk Development Team
What is the best way to disable rtp and jitter information from debugging,
Dan Cropp
Encountered a crash, what is best way to tell if it has been fixed or now,
Dan Cropp
Subscribing to events on AMI login,
TTT
Can ShanSpy be used on Local Channels?,
Carlos Chavez
Parallel dialoog with different Alert-Info headers,
Dirk-Willem van Gulik
AEAP experience,
marek
Media flow between them,
Jerry Geis
Asterisk Release 20.4.0,
Asterisk Development Team
Asterisk Release 18.19.0,
Asterisk Development Team
audio from soft phone actual phone from cloud,
Jerry Geis
memory leak?,
Federico
Get manager user info after AMI authentication,
TTT
Is there a good Python library for AMI?,
Carlos Chavez
Manager permissions for CoreSettings command,
TTT
AMI versions,
TTT
asterisk sees private IP address of a device behind NAT,
Fourhundred Thecat
Is there a way to compile app_macro in 16.30.1,
Federico
Asterisk Release 20.3.1,
Asterisk Development Team
Asterisk Release certified-18.9-cert5,
Asterisk Development Team
Asterisk Release 19.8.1,
Asterisk Development Team
Asterisk Release 18.18.1,
Asterisk Development Team
Asterisk Release 16.30.1,
Asterisk Development Team
problem getting dahdi-linux to work with kernel 6.1.0-10,
John Covici
Re: Getvar of CHANNEL not working for a couple of items,
TTT
Getvar of CHANNEL not working for a couple of items,
TTT
SetCallerPres command gone,
TTT
AGI script commands,
TTT
Setting codec on originating (calling) channel with chan_pjsip (SIP_CODEC_INBOUND equivalent),
Michael Ulitskiy
Issue with PJSIP contacts being "unavailable",
asterisk
Get channel variables via ARI/AMI,
TTT
Why is WebRTC treated differently from regular SIP in Asterisk,
TTT
WebRTC signaling,
TTT
PMS integration,
Jeff LaCoursiere
Asterisk not replacing private FROM ip with public IP in INVITE,
TTT
PJSIP not performing outbound authentication,
TTT
Multiple phones on same PJSIP account,
TTT
Get SIP Call-ID from ARI,
TTT
Expanding my answering-machine system,
Steve Matzura
Add user to conference via ReST/ARI,
TTT
Event showing who called who,
TTT
Problem with pjsip,
Yves
Adding Voicemail to My System,
Steve Matzura
Listen to ARI events,
TTT
Can't stop Mixmonitor,
Jon Bonilla (Manwe)
Question on ring count on incoming circuits,
Steve Matzura
A stupid problem with Playback,
Steve Matzura
Problems solved,
Steve Matzura
Function DENOISE not registered,
Fourhundred Thecat
Re: Problems Solved, Two Remaining,
Steve Matzura
Problems Solved, two left,
Steve Matzura
Message not available
Message not available
Re: Problems Solved, two left,
Daryl Richards
Problems with inbound connection and registering phone,
Steve Matzura
Asterisk Release 20.3.0,
Asterisk Development Team
Asterisk Release 18.18.0,
Asterisk Development Team
Ready to throw up my hands in defeat,
Steve Matzura
SAY_DTMF_INTERRUPT not working,
Dovid Bender
asterisk 18.17.1 unreachable,
Jerry Geis
Calls running forever / CDRs inaccurate,
Markus
Opus: No translation path after upgrade ubuntu focal => jammy,
Benoît Panizzon
DUNDI anyone?,
Benoit Panizzon
Broken link in LICENSE file,
John Runyon
Compiling asterisk makes Systemd timeout when starting the service,
Federico
Asterisk issue reporting is now live on GitHub,
Asterisk Development Team
Asterisk translates 200 OK + SDP into 488 not acceptable here after both side agreed on codec.,
Benoît Panizzon
Reminder: Issues and Code Contribution move to GitHub,
Asterisk Development Team
ODBC Crash,
Federico
Source code for AGI GET DATA command,
Rhys Hanrahan
Asterisk Infrastructure Move to GitHub,
George Joseph
couldn't allocate a port for RTP instance,
Fourhundred Thecat
Setting PJSIP header from AMI,
Alex Zarubin
TLS and NAT,
Steve Matzura
Remote-Party-ID set to 0 on re-invite using pjsip in Asterisk 16.,
Steve Sether
Intro and question,
Steve Matzura
log custom variable in cdr,
Fourhundred Thecat
Asterisk 20.2.1 Now Available,
Asterisk Development Team
Asterisk 18.17.1 Now Available,
Asterisk Development Team
401 error,
Jerry Geis
Asterisk 20.2.0 Now Available,
Asterisk Development Team
Asterisk 18.17.0 Now Available,
Asterisk Development Team
cdr_sqlite3,
Fourhundred Thecat
Mailing Lists,
Joshua C. Colp
Asterisk PJSIP setting don't fragment bit on UDP,
Benoit Panizzon
5s delays before executing the dialplan,
Kingsley Tart
RTP address learning and timing problem,
David Cunningham
Asterisk simply stops call processing,
Antony Stone
Not reporting IP of the incoming connection 18.14.0,
Jerry Geis
github - mlan,
Jeff LaCoursiere
Asterisk rtp.conf stunaddr setting - what happens if there is an outage,
Dan Cropp
set codec based on B side,
Fabian Borot
Is there a list of Channel ARI requests that are allowed when the call is not handed off to the Stasis application,
Dan Cropp
Re: asterisk-users Digest, Vol 221, Issue 2,
Ron Lockard
Question on ARI externalMedia,
Dan Cropp
Testing,
Joshua C. Colp
Certified Asterisk 18.9-cert4 Now Available,
Asterisk Development Team
mailing list working?,
marek
sip trunk, parsing DID,
Marc SCHAEFER
Global variables in global variables,
Antony Stone
sender IP of unwanted SIP user,
astuserlist
Dahdi Compile on 22.04 LTS,
Jerry Geis
PlayBack,
astuserlist
monitor files gsm format split,
astuserlist
Two calls from same server to end device,
Jerry Geis
Asterisk 18.12.1 to 18.15.0 upgrade seems to have introduced a behavior where PJSIP is unable to send a response to OPTIONS (seems to resolve after anywhere a period of time),
Dan Cropp
Receive DTMF and record audio at the same time,
Rhys Hanrahan
cannot load res_geolocation.so,
Nick Olsen
Codec opus returned invalid number of samples,
Fourhundred Thecat
Upgraded from asterisk 18.14.0 to 20.0.0 and inbound registration(?) is now failing,
Justin Piszcz
Asterisk 16.29.1, 18.15.1, 19.7.1, 20.0.1 Now Available,
Asterisk Development Team
G.729 Annex B or AB support in Asterisk,
Tahir Almas Dhesi
Asterisk unable to do DNS lookups,
TTT
possibility to cancel call duration limit set in app Dial with options S(x) or L(x:y:z) during a call,
Nenad Radosavljevic
MixMonitor not recording through transfer,
Carlos Chavez
Handling SIP refers when using a SIP Proxy,
Dovid Bender
Voicemail Transcription with openai/whisper,
Doug Lytle
Re: Voicemail Transcription with openai/whisper,
David Rebarchik
cps limit of asterisk,
Tahir Almas Dhesi
is KTLS usable with asterisk+libsrtp?,
marek
Force voicemail check / MWI generation,
Nick Olsen
Answer()ing a local Originate takes 500ms!?,
Antony Stone
Asterisk 18.14.0 console dsp,
Jerry Geis
asterisk kernel crash,
Jerry Geis
menuselecting res_corosync,
John Harragin
CSeq reset on re-INVITE,
David Cunningham
Confbridge for 80 devices,
Jerry Geis
Asterisk 20.0.0 Now Available,
Asterisk Development Team
Asterisk 19.7.0 Now Available,
Asterisk Development Team
Asterisk 18.15.0 Now Available,
Asterisk Development Team
Asterisk 16.29.0 Now Available,
Asterisk Development Team
RTP audio,
Jerry Geis
@-sign gets transmitted as %40 in outgoing SIP packets (CallerID),
Markus
asterisk 18.14.0 connected to Call Manager,
Jerry Geis
pjsip endpoint reachable,
marek
Muliticast not connecting,
Jerry Geis
Trying asterisk on AWS,
Jerry Geis
asterisk 8.14.0 and multicast sometimes not hear anythign,
Jerry Geis
VoiceMail() stop dialplan processing,
Antony Stone
libpri compile ubuntu 22.04,
Jerry Geis
Two quick questions,
Jerry Geis
res_http_media_cache curl Options from Config,
Jöran Vinzens
What conditions require the AMI_VERSION number to be bumped?,
Dan Cropp
Use of TONE_DETECT to detect dial tone in call (Asterisk18),
Stephen Moran
Problems detecting hangup after hold / resume,
Antony Stone
parsing P-Asserted-Identity if Privacy: id,
marek
Channel names with semicolons (sending again),
Antony Stone
Re: Channel names with semicolons,
Antony Stone
Question on Originate with EarlyMedia,
Dan Cropp
AEL switch & case,
Antony Stone
Forward incoming call to recipients.,
Ken D'Ambrosio
Originate with label?,
Antony Stone
I think there may be a bug in 18.14.0 ${GEOLOC_PROFILE(profile_precedence)}, seems to always return prefer_incoming,
Dan Cropp
Asterisk 16.16.1 crash upon receiving image 0 udptl t38 sdp,
Benoit Panizzon
extensions.conf [General] settings,
Antony Stone
Multicast on asterisk 13.30.0 weird issue.,
Jerry Geis
Asterisk 19.6.0 Now Available,
Asterisk Development Team
Asterisk 18.14.0 Now Available,
Asterisk Development Team
Asterisk 16.28.0 Now Available,
Asterisk Development Team
Cannot send faxes,
Luca Bertoncello
blogs.asterisk.org broken,
Michael Maier
Question on resources,
Jerry Geis
Run asterisk -rx "command" and get plain text output,
Carlos Chavez
Geo location 18.14.0-rc1 question,
Dan Cropp
parallel dial problem (used to work on Asterisk 13),
Kingsley Tart
Asterisk "we couldn't allocate a port for RTP" errors,
David Cunningham
Question about the Geo Location support being added,
Dan Cropp
DUNDi peers disconnect after being connected for months or years, cannot reconnect again,
Court Campbell
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