Asterisk Internet Phone System
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- STRFTIME get always 0 milliseconds asterisk 13,
Raimundo Pérez Nieves
- Is order of channels shown by Function_CHANNELS consistently newest first?,
Jonathan H
- How to force Asterisk to reply with floating IP with chan_sip ?,
Olivier
- pjsip aor stays in status created,
marek cervenka
- SIPp scenario file for testing UAC Authentication with Asterisk ?,
Olivier
- How can I connect an existing Confbridge to a new SIP channel when DIALEDPEERNAME is empty?,
Jonathan H
- Missing audio on playback in 16.0,
Karsten Wemheuer
- AMI not listening on secondary IP address?,
Antony Stone
- Finding out if channels is up,
Dovid Bender
- How best to run a SIPp test on a remote host,
Olivier
- After updating to 16 "Some non-required modules failed to load",
Jonathan H
- Connecting an existing conference via PJSIP?,
Jonathan H
- Asterisk 15 and Cepstral,
Carlos Chavez
- Re: Is there any way to pass caller id to,
Ivan Demkovitch
- Re: Is there any way to pass caller id to cell phone,
Daniel Friedman
- Re: Is there any way to pass caller id to cell phone?,
Eric Klein
- Disabling a trunk at runtime,
Telium Support Group
- asterisk 16 manager --END COMMAND--,
Dmitry Melekhov
- Is there any way to pass caller id to cell phone?,
Ivan Demkovitch
- What's the best way of extracting call data which has been written to flat files?,
Jonathan H
- How to defer SDP in ACK for unit testing purposes,
Olivier
- Asterisk 16.0.0 Now Available,
Asterisk Development Team
- Makefile target to generate asterisk.service file,
Olivier
- First attempt with statsd,
Olivier
- Explain module reloading error message,
Olivier
- Dropped calls when all DAHDI lines in use,
Andrew Martin
- CURL to post application/json (David P),
Stefan Viljoen
- Spontaneous reboot due to MySQL lookups ? (Jonas Kellens),
Stefan Viljoen
- CURL to post application/json,
David P
- Spontaneous reboot due to MySQL lookups ?,
Jonas Kellens
- Any idea what causes "Oooh, got a frame with format of g729 on channel 'PJSIP/121-000001d2' when we're sending 'ulaw', switching to match",
Dan Cropp
- Non-matching linkedid on CDR Records [SEC=UNCLASSIFIED],
Calum Power
- Call Queue Data,
Tech Support
- messagesend to SIP peer in sip.conf (or otherwise authenticated),
Brian J. Murrell
- Stop,
Karen York
- chan_pjsip: DTMF mode "auto_info" on endpoints,
Floimair Florian
- WebRTC as Softphone substitute ?,
Olivier
- Asterisk 15.6.1. Symbol pjsip_tls_transport_start2 not found,
Dmitriy Serov
- Re: Convert SIP to PJSIP,
John T. Bittner
- AST-2018-009: Remote crash vulnerability in HTTP websocket upgrade,
Asterisk Security Team
- Asterisk 13.23.1, 14.7.8, 15.6.1 and 13.21-cert3 Now Available (Security),
Asterisk Development Team
- AES-67,
Jerry Geis
- Asking,
modou lo
- IVR call simulation on Asterisk 15 server,
Priyaranjan Nayak
- AGI timeout option,
Patrick Wakano
- Is it possible to retrieve header fields from a SIP UPDATE packet?,
Dan Cropp
- hangup the _called_ channel ?,
sean darcy
- Can someone provide some insight on WebRTC vs a generic SIP library in a browser?,
Dan Cropp
- failed to find existing extension,
asterisk
- Asterisk 16 AMI changes,
Telium Support Group
- Asterisk 15.6.0 Now Available,
Asterisk Development Team
- Asterisk 13.23.0 Now Available,
Asterisk Development Team
- Voicemail help when listening to messages,
Nathan Ward
- STUN re-evalutation every 2 minutes ??,
sean darcy
- Community forum ?,
sean darcy
- 401 unauthorized,
Jerry Geis
- getting invites to rtp ports ??,
sean darcy
- feeling n00b again,
asterisk
- Call pickup on channel sip with SNOM phones issue,
Hans-Peter Jansen
- Re: asterisk-users Digest, Vol 168, Issue 14,
Ahmed Chohan
- How do I retrieve the Call-ID from the SIP INVITE when using Originate on PJSIP?,
Dan Cropp
- Merging 2 conference bridges,
Ahmed Chohan
- change dialing process on live call,
Khalil Khamlichi
- jump on DTMF while MP3Player is on,
Saint Michael
- How to implement an ENUM mock database ?,
Olivier
- Encrypting passwords in config files,
Floimair Florian
- Issues with install DAHDI,
Dovid Bender
- ContactStatus AMI Event on PJSIP Reregistration,
Joshua Colp
- Is there a way to remove launching shell command from Asterisk CLI,
Olivier
- Re: How to properly execute rasterisk over SSH ? [SOLVED],
Olivier
- How to properly execute rasterisk over SSH ?,
Olivier
- Scaling voicemail,
Nathan Ward
- pjsip: TOS not working any more,
Michael Maier
- Disable asterisk ssl how to,
Saint Michael
- Queue breaks Dynamic_Features on Attended Transfer,
Daniel Journo
- PJSIP redirect_method=uri_core and header modifications,
Daniel Tryba
- Re: Asterisk 13.22.0 - No channel type registered for 'Agent' when queue rings - solved,
Stefan Viljoen
- Struggling to make sense of sending DTMF and why DIAL is trying to make multiple calls?,
Jonathan H
- Asterisk 13.22.0 - No channel type registered for 'Agent' when queue rings,
Stefan Viljoen
- 400 reply to INVITE not properly treated,
Patrick Wakano
- Asterisk 11 with volume control for Confbridge,
Jerry Geis
- Re: asterisk-users Digest, Vol 167, Issue 21,
Raimundo Pérez Nieves
- Increasing timeout before ending call from AMI,
Raimundo Pérez Nieves
- Re: asterisk-users Digest, Vol 167, Issue 17,
Stefan Viljoen
- Any way of "flattening out" 2 channels back into one?,
Jonathan H
- SRV with pjsip on Asterisk 15.5: yes or no?,
Jonathan H
- dialplan reload not showing debug info even with debug on (ast 15.5),
Jonathan H
- Can someone please help with this sip2sip pjsip_wizard "no matching endpoint" issue?,
Jonathan H
- Asterisk indication tones per channel,
Sam Basan
- Re: Asterisk 13 - system() dialplan app cannot call bash scripts,
Stefan Viljoen
- Re: SHELL() function Asterisk 13 - can only accept one paramter in string?,
Stefan Viljoen
- SHELL() function Asterisk 13 - can only accept one paramter in string?,
Stefan Viljoen
- Edit: Asterisk 13.22.0 - "stat" dialplan function clears channel vars?,
Stefan Viljoen
- Asterisk 13.22.0 - "stat" dialplan function clears channel vars?,
Stefan Viljoen
- How to know the IP of "manager show connected" in dialplan,
Saint Michael
- Re: [asterisk-app-dev] how to use snoopChannel,
Joshua Colp
G729 (Dmitry Melekhov),
Saint Michael
Segfault on libasteriskpj.so.2,
Carlos Chavez
Re: [asterisk-app-dev] AGI stream audio from URI,
Matthew Jordan
No audio on direct call from trunk to SPA-8000,
Carlos Chavez
Asterisk pjsip realtime extensions,
Benjamin Marty
Recompiling Ast results in a binary with differing SHA256 sums?,
Stefan Viljoen
Withholding Answer Supervision,
Dovid Bender
Asterisk 15.5.0 Now Available,
Asterisk Development Team
Asterisk 13.22.0 Now Available,
Asterisk Development Team
How to steal an answered call?,
David Cunningham
MixMonitor multiple times to the same file,
Patrick Wakano
Passing arguments to the 'mailcmd' option in voicemail.conf,
Tech Support
MixMonitor and ChanSpy whisper,
Patrick Wakano
Video call recording and mobile push notifications,
Jeremy Renner
A survey on Asterisk-based call-centres - Help needed,
Lenz Emilitri
Core show channels concise = deprecated,
Telium Support Group
rtp port and strictrtp,
Dovid Bender
pri_check_event returned error 22 (Invalid argument),
陈杨文
Button for call forward and button for pickup call of another extension,
bilal ghayyad
Busy indicator for FXO line or extension,
bilal ghayyad
Asterisk crashing on AAAA lookup,
Dovid Bender
Asterisk not matching longest prefix with include,
Dovid Bender
Best way to update ever changing dialplans,
Dovid Bender
GSM card or GSM adaptor?,
bilal ghayyad
Recommended Linux version or how to compile DAHDI on Fedora?,
Ira
Voicemail Directory,
Doug Lytle
Asterisk receiving 415 Unsupported Media Type upon T.38 invite behaving absolutely weird.,
Benoit Panizzon
Do you set chan_sip's ignoresdpversion to true ?,
Olivier
Asterisk Realtime PJSIP - slow output on "pjsip show xxxxx" commands,
Floimair Florian
Only 8kHz recorded after disallowing all but G722 codec on inbound,
David P
[Asterisk-video] (no subject),
Pankaj Pandey
How to ignore REFER entirely with chan_sip or PJSIP ?,
Olivier
AMD min amount of words,
Dovid Bender
MixMonitor recording when in the holding bridge,
Patrick Wakano
AST-2018-008: PJSIP endpoint presence disclosure when using ACL,
Asterisk Security Team
AST-2018-007: Infinite loop when reading iostreams,
Asterisk Security Team
Asterisk 15.4.1, 13.21.1, 14.7.7, 13.18-cert4 and 13.21-cert2 Now Available (Security),
Asterisk Development Team
Start audio call and enable video later,
Stefan Tichy
Head request with curl in Asterisk,
Dovid Bender
getting real sip status after dial,
Khalil Khamlichi
Asterisk kafka connector feedback,
Alex Pappas
T-38 re-invite issue,
D'Arcy Cain
AMI manager logins - omitting from logging output?,
Antony Stone
Documentation for media caching,
Dovid Bender
Function CHANNELS,
Matt Hamilton
Using ControlPlayback with AWS S3,
Dovid Bender
pjsip doesn't function,
Marko Tirs
Certified Asterisk 13.21-cert1 Now Available,
Asterisk Development Team
Questions about SIP From, P-Asserted-Id fields and Diversion headers ?,
Olivier
How to execute priorities following a caller hangup in a successful Dial?,
David P
shell dialplan application blocking,
Benjamin Marty
doing dnsmgr_lookup for,
Jonas Kellens
Queue of automated members,
David P
Long extensions that contain dashes,
David P
Dial to FastAGI application appears as 1-second CDR - how do I fix?,
Alex Villacís Lasso
Asterisk TLS 5061 not listening,
Benjamin Marty
Big leap - 1.8 to 15.4.0,
Stefan Viljoen
Trying to add MoH to conference bridge,
Mike Diehl
New Dahdi,
Jerry Geis
Testing for real from a non-digium email,
Matthew Fredrickson
Testing...,
Matt Fredrickson
testing users list,
Matt Ball
Test from Digium address,
Matt Fredrickson
test delivery for lists.digium.com,
Brad Burns
asterisk rules,
Matt Ball
One more test,
Matt Fredrickson
More testing,
Matt Fredrickson
IBM Watson Voicemail Transcription,
Doug Lytle
Looking for better fax handling,
D'Arcy Cain
Re: res_fax_spandsp - information about used protocol t38 or g711?,
Andre Gronwald
setting contact within asterisk -rx 'channel originate local ...',
Andre Gronwald
res_fax_spandsp - information about used protocol t38 or g711?,
Michael Maier
AMI status events with res_fax_spandsp.so,
Steven Wheeler
Decoding SIP register hack,
sean darcy
Streaming MoH from iHeart radio?,
Mike Diehl
When should a Progress or Ringing be used in a today's telephony ?,
Olivier
SIP Codec negotiation,
Steve Edwards
PJSIP multiple Unlimitel accounts,
dbc_asterisk
Sound files,
Dovid Bender
multi step auth?,
Jeff LaCoursiere
Reject call from Asterisk dialplan,
Mike
Passing parameter to Queue-called macro,
Stefan Viljoen
7965G sporadically not able to make calls via chan_sip,
John Kinsner
Asterisk 15.4.0 Now Available,
Asterisk Development Team
Asterisk 13.21.0 Now Available,
Asterisk Development Team
DTMF tones in MixMonitor recording,
Patrick Wakano
AstriCon Approaching, Super Earlybird Pricing Expires In 3 Days,
Matt Fredrickson
Question on PJSIP's endpoint section in wiki,
Olivier
Re: PJSIP global section ignored in Asterisk 13.14.1 [SOLVED],
Olivier
PJSIP global section ignored in Asterisk 13.14.1,
Olivier
How to check modules loading order or force such order ?,
Olivier
Explain PJSIP user matching within inbound SIP trunks,
Olivier
headers in master.csv,
John Tuxies
Wanted: WebRTC tutorial,
Bruce Ferrell
Asterisk Community Services Outages,
Matt Fredrickson
Alias for country in indications.conf,
Patrick Wakano
VMWare guest crash with ^@ character in logs. Where to look at to find root cause ?,
Olivier
Re: Possibility to access PJSIP variables from dialplan,
George Joseph
Re: PJSIP error No auth credentials for realm(s) 'asterisk' in challenge,
Joshua Colp
Disable blind and attended transfer during call,
Andrzej Nowrot
Digium IP Phones UNREACHABLE after registration,
Hermann Wecke
Pass through registration / proxy,
Telium Technical Support
withheld caller id,
Atux Atux
Asterisk behind NAT Early Media Video,
Benjamin Marty
PJSip CallerID Question,
Brent Davidson
Re: Asterisk / PRI and Outbound Overlap Dialing (Mtt Cannon),
Mc GRATH Ricardo
Asterisk / PRI and Outbound Overlap Dialing,
Mtt Cannon
Iridium integration / gateway,
Jean-Denis Girard
Strange problem with PRI on 64-bit?,
Tony Mountifield
Asterisk Local channel Earlymedia,
Benjamin Marty
pjsip trunk config question + DNS related error messages,
Kevin Long
Setting outgoing CALLERID without changing CDR(src),
Carlos Chavez
IFTIME and timezones,
martin f krafft
Sorry for interruption of service,
Matt Fredrickson
More testing - sorry guys,
Matt Fredrickson
Looking for C library for the Asterisk AMI,
Tech Support
h264 recording,
Benjamin Marty
Audio Dropouts During Call,
Brent Davidson
Re: Client Asterisks can't connect when main Asterisk reboot,
Antony Stone
Re: AMI potential memory leak which may be causing a crash,
Dan Cropp
Asterisk as gateway,
Atux Atux
Voip realtime analysis tools,
Marcus Kvarsell
AMI potential memory leak,
Dan Cropp
invite to conference by a call file,
Atux Atux
Is 100 trying mandatory? Can asterisk answer with 180 without prior 100 trying?,
Benoit Panizzon
Asterisk 15.3.0 Now Available,
Asterisk Development Team
Asterisk 13.20.0 Now Available,
Asterisk Development Team
PJSIP Originate,
Dan Cropp
What is ASTDB /pbx/UUID for ? Can I duplicate whole ASTDB from cluster active member to passive member ?,
Olivier
Problems with app_cdr writing CDRs nowhere,
Mike
DIALSTATUS vs HANGUPCAUSE,
Patrick Wakano
Getting DTMF from Asterisk Record?,
David Cunningham
Bank holidays read from file?,
Atux Atux
Zombie PJSip Channel,
Brent Davidson
Database re-connect issue,
D'Arcy Cain
[OT] Load testing with SIPp,
Olivier
Comparison of PJSIP and SIP in Asterisk database,
Olivier
Half Off Topic Questions,
Markus Weiler
[Index of Archives]
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[Asterisk App Development]
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[Yosemite Campsites]
[IETF Sipping]
[Asterisk Books]