Re: [asterisk-users] HELP! AGI AUTOHANGUP does not seem to hangup the channel.

2021-05-27 Thread Steve Edwards
er you 'set autohangup x' just set 'TIMEOUT(absolute)=${EPOCH}+x.' -- Thanks in advance, --------- Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/

Re: [asterisk-users] HELP! AGI AUTOHANGUP does not seem to hangup the channel.

2021-05-26 Thread Steve Edwards
> agi_version: 13.14.1~dfsg-2+deb9u4 AGI Tx >> agi_callerid: 55 AGI Tx >> agi_calleridname: Steve Edwards AGI Tx >> agi_callingpres: 0 AGI Tx >> agi_callingani2: 0 AGI Tx >> agi_callington: 0 AGI Tx >> agi_callingtns: 0 AGI Tx >> agi_dnid: * AGI T

Re: [asterisk-users] AGI: Why is stream file and wait for digit result ASCII, but get data is "normal"?

2021-05-24 Thread Steve Edwards
789' -- Thanks in advance, ----- Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281 -- _ -- Bandwidth and C

Re: [asterisk-users] S3 Bucket support for playing sound files

2021-05-06 Thread Steve Edwards
On Thu, 6 May 2021, Jonathan H wrote: "bumps up the outgoing volume to +7" I use 'normalize --amplitude=-22dB" to adjust volume levels to consistent levels. -- Thanks in advance, --------- Steve Edwa

Re: [asterisk-users] Loading Json values into asterisk as variable values

2021-02-26 Thread Steve Edwards
in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281-- _ -- Bandwidth

Re: [asterisk-users] Loading Json values into asterisk as variable values

2021-02-25 Thread Steve Edwards
. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281

Re: [asterisk-users] AGI Script Returning 4

2021-01-30 Thread Steve Edwards
(). Almost always exit(). -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281

Re: [asterisk-users] Get a SHAKEN Identity Token (Alexander Perkins)

2021-01-25 Thread Steve Edwards
out of service copper number, 555-555-. I'm all for the discussion, but can you start a new thread so we don't keep associating the innocent party (the OP) with this spammer. -- Thanks in advance, - Steve Edwards

Re: [asterisk-users] Get a SHAKEN Identity Token (Alexander Perkins)

2021-01-24 Thread Steve Edwards
---- Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check o

Re: [asterisk-users] DAHDI timing

2021-01-06 Thread Steve Edwards
xes using cards so I can't test.) -- Thanks in advance, ----- Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244

Re: [asterisk-users] DAHDI timing

2021-01-06 Thread Steve Edwards
est Attempting to test a timer with 50 ticks per second. Using the 'timerfd' timing module for this test. It has been 1000 milliseconds, and we got 50 timer ticks -- Thanks in advance, ----- Steve Edwards sedwa...@sedwards.

Re: [asterisk-users] Detect if people is talking

2020-12-31 Thread Steve Edwards
in advance, ----- Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281 -- _ -- Bandwidth and Colocation Provid

Re: [asterisk-users] faxdetect timeout configuration

2020-12-29 Thread Steve Edwards
/res_pjsip/pjsip_configuration.c -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281

Re: [asterisk-users] HELP! I can't get my Cisco CP-7960G IP hardphone to register on my Asterisk VoIP IP PBX SIP Server with FreePBX GUI

2020-12-23 Thread Steve Edwards
only allowed up to 31 character passwords. You may find it useful to use tcpdump with '-w' to write the packets to a file and then analyze with sngrep. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com

Re: [asterisk-users] how do I run a command on "Failed to authenticate" ?

2020-09-11 Thread Steve Edwards
---- Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the ne

Re: [asterisk-users] Stir Shaken is upon us

2020-07-15 Thread Steve Edwards
On Sun, 12 Jul 2020, Steve Edwards wrote: So this is a provider issue, not an end user issue and 'June 30, 2021' doesn't sound like 'soon.' If this is legit, why haven't my providers said squat? Seems one of my providers, Vitelity (iax.cc to us old timers), when asked, is not panicking

Re: [asterisk-users] Stir Shaken

2020-07-13 Thread Steve Edwards
is 'token' add any value? What am I missing? -- Thanks in advance, ----- Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwar

Re: [asterisk-users] Stir Shaken is upon us

2020-07-12 Thread Steve Edwards
-- Thanks in advance, --------- Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -

Re: [asterisk-users] Redis in place of astdb

2020-07-08 Thread Steve Edwards
, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new

Re: [asterisk-users] Redis in place of astdb

2020-07-08 Thread Steve Edwards
, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281 -- _ -- Bandwidth and Colocation Provided

Re: [asterisk-users] Forbidden call

2020-06-27 Thread Steve Edwards
On Fri, 12 Jun 2020, Jerry Geis wrote: Any chance you can configure the speaker to syslog to your host so you may get a clue why the speaker is rejecting? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com

Re: [asterisk-users] Any api (agi/ari/ami) equivalent of "core show calls"?

2020-06-14 Thread Steve Edwards
in advance, ----- Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281 -- _ -- Bandwidth and

Re: [asterisk-users] Any api (agi/ari/ami) equivalent of "core show calls"?

2020-06-13 Thread Steve Edwards
as the caller hangs up), and then  rewrite and reload again when there's a new caller. How about ARA to configure MOH and then just update the database. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice

Re: [asterisk-users] Forbidden call

2020-06-11 Thread Steve Edwards
ly from the Ethernet on the speaker to a NIC on the computer? It doesn't matter, just curious :) The only thing that will tell you what is going on is the packets. Crank up 'sip set debug on' and see if that yields a clue. -- Thanks in advance, -------

Re: [asterisk-users] problem with logger: syslog vs. file

2020-06-03 Thread Steve Edwards
On Wed, 3 Jun 2020, Fourhundred Thecat wrote: On 2020-06-03 17:21, Steve Edwards wrote: How about:     syslog.local0   = error,verbose,warning no debugging detail.     syslog.local0   = debug,error,verbose,warning include debugging detail

Re: [asterisk-users] problem with logger: syslog vs. file

2020-06-03 Thread Steve Edwards
about: syslog.local0 = error,verbose,warning no debugging detail. syslog.local0 = debug,error,verbose,warning include debugging detail. -- Thanks in advance, - Steve E

Re: [asterisk-users] STIR-Shaken

2020-05-28 Thread Steve Edwards
, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281 -- _ -- Bandwidth and Colocation Provided by http://www.api

Re: [asterisk-users] Stir-Shaken for asterisk

2020-05-27 Thread Steve Edwards
On Wed, 27 May 2020, Saint Michael wrote: We are in the business of... Then this probably should have been posted on -biz. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST

Re: [asterisk-users] rotatestrategy = none not working

2020-05-20 Thread Steve Edwards
ne") == 0) { rotatestrategy = NONE; } else { fprintf(stderr, "Unknown rotatestrategy: %s\n", s); } So, backport or upgrade? Also, inquiring minds want to know why the enum is in powers of 2? It's not like we can set

Re: [asterisk-users] rotatestrategy = none not working

2020-05-20 Thread Steve Edwards
in advance, ----- Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281 -- _ --

Re: [asterisk-users] rotatestrategy = none not working

2020-05-20 Thread Steve Edwards
---- Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the ne

Re: [asterisk-users] Perl AGI: read variable with quotes

2020-01-24 Thread Steve Edwards
On Fri, 24 Jan 2020, Steve Edwards wrote: 2) How about doing 'GET FULL VARIABLE' in your Perl script? Sorry. After a couple more cups of tea I think this was a bit vague. Try whatever call/method in your library that does 'GET FULL VARIABLE' on '${PJSIP_HEADER(read,P-Asserted-Identity

Re: [asterisk-users] Perl AGI: read variable with quotes

2020-01-24 Thread Steve Edwards
ll header? Try 'verbose(PAI = ${PAI})' or something similar. 2) How about doing 'GET FULL VARIABLE' in your Perl script? You can set the channel variable PAI in the AGI if needed back in the dialplan. -- Thanks in advance, -------

Re: [asterisk-users] USB dahdi fxo ?

2019-12-13 Thread Steve Edwards
Sipura 3000? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281

Re: [asterisk-users] Two sip extensions

2019-07-19 Thread Steve Edwards
, increase consistency, and reduce maintenance. -- Thanks in advance, ----- Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwar

Re: [asterisk-users] Two sip extensions

2019-07-18 Thread Steve Edwards
because I'm just that kind of guy :) -- Thanks in advance, --------- Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve

Re: [asterisk-users] Find out which key ended recording?

2019-06-07 Thread Steve Edwards
On Fri, 7 Jun 2019, David Cunningham wrote: We're using Perl and so far I haven't found an equivalent there. On Thu, 6 Jun 2019, Steve Edwards wrote: I'm not much of a Perl programmer... But you should never turn down an opportunity to develop your skills :) Try something like

Re: [asterisk-users] Find out which key ended recording?

2019-06-07 Thread Steve Edwards
Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4

Re: [asterisk-users] Find out which key ended recording?

2019-06-07 Thread Steve Edwards
On Fri, 7 Jun 2019, David Cunningham wrote: We're using Perl and so far I haven't found an equivalent there. On Thu, 6 Jun 2019, Steve Edwards wrote: I'm not much of a Perl programmer... But you should never turn down an opportunity to develop your skills :) Try something like

Re: [asterisk-users] Find out which key ended recording?

2019-06-06 Thread Steve Edwards
} Looks like agi_environment.result is your Huckleberry. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244

Re: [asterisk-users] Play Music While Processing AGI Script

2019-05-14 Thread Steve Edwards
ful (long) options and are not dependent upon passing arguments in a particular order. -- Thanks in advance, ----- Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/stev

Re: [asterisk-users] Sending SMS and SIM card

2019-04-23 Thread Steve Edwards
in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281

Re: [asterisk-users] Forking AGI or GoSub

2019-04-10 Thread Steve Edwards
. The only caveat is to not interact (stdin/stdout) with Asterisk until 'stream file' in the thread completed. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST

Re: [asterisk-users] AMI mulitple calls quickly

2019-03-12 Thread Steve Edwards
, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com

Re: [asterisk-users] Outbound caller ID ignored

2019-01-13 Thread Steve Edwards
is not a POTS feature. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281

Re: [asterisk-users] Custom langagues

2019-01-01 Thread Steve Edwards
---- Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281-- _ -- Bandwidth and Colocation Provided by http://

Re: [asterisk-users] Capture SIP all the time

2018-12-05 Thread Steve Edwards
somebody says 'hey, my call didn't connect yesterday' I have something to work with. sngrep is a great tool for searching for calls and displaying decoded dialogs. -- Thanks in advance, - Steve Edwards sedwa

Re: [asterisk-users] Best way to update ever changing dialplans

2018-06-25 Thread Steve Edwards
, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281 -- _ -- Bandwidth and Colocation Provided by http

Re: [asterisk-users] Decoding SIP register hack

2018-05-17 Thread Steve Edwards
? A clever solution to a mobile user base is to use knockd to allow remote access. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards

Re: [asterisk-users] SIP Codec negotiation

2018-05-17 Thread Steve Edwards
On Fri, May 11, 2018, at 10:36 AM, Steve Edwards wrote: So, Asterisk will defer it's choice of codec to match the codec it detects in the incoming stream? On Fri, 11 May 2018, Joshua Colp wrote: It depends on the channel driver and configuration. The chan_sip module always matching

Re: [asterisk-users] SIP Codec negotiation

2018-05-11 Thread Steve Edwards
, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281 -- _ -- Bandwidth and Colocation Provided by http

Re: [asterisk-users] SIP Codec negotiation

2018-05-11 Thread Steve Edwards
On Thu, May 10, 2018 at 11:44:14AM -0700, Steve Edwards wrote: I receive an INVITE/SDP containing: m=audio 11310 RTP/AVP 3 0 101 which I interpret as gsm, ulaw, rfc2833. and I reply with an OK/SDP containing: m=audio 15884 RTP/AVP 0 3 101 which I interpret as ulaw, gsm

[asterisk-users] SIP Codec negotiation

2018-05-10 Thread Steve Edwards
? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281

Re: [asterisk-users] AGI fails bad permission

2018-02-23 Thread Steve Edwards
/adddnc.php 7) Check the 'r' and 'x' bits on /var/, /var/lib/, /var/lib/asterisk/, /var/lib/asterisk/agi-bin/. 8) cat /var/lib/asterisk/agi-bin/adddnc.php -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com

Re: [asterisk-users] [OT] How to use audio files with SIPp

2018-02-09 Thread Steve Edwards
On Fri, 9 Feb 2018, Olivier wrote: 3. How do you capture an RTP flux with thark or tcpdump ? Take a look at 'pcapsipdump.' -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST

Re: [asterisk-users] Answered time on channel

2018-01-02 Thread Steve Edwards
On Tue, 26 Dec 2017, Eric Wieling wrote: Don't use an 'h' extension, use a hangup handler.   On 12/26/2017 04:43 PM, Steve Edwards wrote: Why? On Tue, 2 Jan 2018, Eric Wieling wrote: From the hangup handler specification: Hangup handlers are an alternative

Re: [asterisk-users] Answered time on channel

2017-12-26 Thread Steve Edwards
On Tue, 26 Dec 2017, Eric Wieling wrote: Don't use an 'h' extension, use a hangup handler.   Why? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https

Re: [asterisk-users] Chan Local, Originate and slin

2017-11-22 Thread Steve Edwards
, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281 -- _ -- Bandwidth and Colocation

Re: [asterisk-users] Chan Local, Originate and slin

2017-11-22 Thread Steve Edwards
ooldir}/outgoing/ and then 'mv' the file to that directory. -- Thanks in advance, --------- Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve

Re: [asterisk-users] Call preemption

2017-11-08 Thread Steve Edwards
' opportunities. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281

Re: [asterisk-users] A bit OT - Configure GoIP for Asterisk

2017-10-03 Thread Steve Edwards
On Mon, 2 Oct 2017, Antony Stone wrote: On Monday 02 October 2017 at 20:58:33, Steve Edwards wrote: I recently received a GoIP-32 for a client project -- primarily outbound calling. How should a GoIP be configured for Asterisk? Have you tried http://www.hybertone.com/en/solutionsClass.asp

[asterisk-users] A bit OT - Configure GoIP for Asterisk

2017-10-02 Thread Steve Edwards
. How did you configure your GoIP and why? What do your relevant sip.conf section(s) look like? What does your dial command look like? So far, all I've got out of it is a 503 Declined. -- Thanks in advance, - Steve Edwards

[asterisk-users] OT: Looking for Kristian Kielhofner document

2017-09-14 Thread Steve Edwards
in advance, ----- Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281 -- _ -- Bandwidth and Colocation Provid

Re: [asterisk-users] ERROR during high volume MoH dialplan

2017-08-31 Thread Steve Edwards
, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281 -- _ -- Bandwidth and Colocation Provided by http://www.api

Re: [asterisk-users] What version of Linux?

2017-08-28 Thread Steve Edwards
x86_64 x86_64 x86_64 sedwards:~$ getconf LONG_BIT 64 -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve

Re: [asterisk-users] AMI column widths

2017-07-07 Thread Steve Edwards
, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281 -- _ -- Bandwidth and Colocation Provided

Re: [asterisk-users] Extensions of sip trunk

2017-06-05 Thread Steve Edwards
n advance, ----- Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281 -- _ -- Bandwidth and Colocation Provided by http:

Re: [asterisk-users] OT: Want to capture all SIP messages

2017-05-31 Thread Steve Edwards
. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281

Re: [asterisk-users] OT: Want to capture all SIP messages

2017-05-31 Thread Steve Edwards
as I change which hosts need monitoring. I know... First world problems :) -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve

Re: [asterisk-users] OT: Want to capture all SIP messages

2017-05-31 Thread Steve Edwards
, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281-- _ -- Bandwidth and Colocation Provided

Re: [asterisk-users] OT: Want to capture all SIP messages

2017-05-31 Thread Steve Edwards
On Wed, 31 May 2017, Daniel Tryba wrote: On Wed, May 31, 2017 at 01:39:25PM -0700, Steve Edwards wrote: What bugs you about the output format? It's been a while, but as I recollect, it included the date/timestamp in the file name of the 'ring buffer' which meant that each time the host

Re: [asterisk-users] OT: Want to capture all SIP messages

2017-05-31 Thread Steve Edwards
On Wed, 31 May 2017, Steve Edwards wrote: I want to capture all SIP messages. I have about 30 hosts in about 6 colos. My first thought was dumpcap, but the output file name format bugs me. What do you use for long term SIP capture? A little more specificity... I'd like the capture

Re: [asterisk-users] OT: Want to capture all SIP messages

2017-05-31 Thread Steve Edwards
On Wed, May 31, 2017 at 12:36:47PM -0700, Steve Edwards wrote: I want to capture all SIP messages. I have about 30 hosts in about 6 colos. My first thought was dumpcap, but the output file name format bugs me. What do you use for long term SIP capture? On Wed, 31 May 2017, Daniel Tryba

[asterisk-users] OT: Want to capture all SIP messages

2017-05-31 Thread Steve Edwards
Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk

Re: [asterisk-users] cmd AGI(), maximum script time.

2017-05-26 Thread Steve Edwards
, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281 -- _ -- Bandwidth

Re: [asterisk-users] How to detect fake CallerID? (8xx?)

2017-05-10 Thread Steve Edwards
to use the 'site landline' to confirm presence -- not their cell phone with the spoofed CID. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com

[asterisk-users] How to detect fake CallerID? (8xx?)

2017-05-10 Thread Steve Edwards
in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281

Re: [asterisk-users] ** in extensions.conf

2017-04-26 Thread Steve Edwards
oly sends the INVITE. -- Thanks in advance, --------- Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.link

Re: [asterisk-users] ** in extensions.conf

2017-04-26 Thread Steve Edwards
me = n, answer() I use an ancient Polycom IP 501 just fine. Does 'dialplan show **@' yield any clues? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST

[asterisk-users] "Your call is not allowed. P U A M I"

2017-04-20 Thread Steve Edwards
, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out

Re: [asterisk-users] Commit dialplan & other config. in memory to disk?

2017-04-07 Thread Steve Edwards
On Thu, 6 Apr 2017, Steve Edwards wrote: You're welcome to the script at: http://www.sedwards.com/recover-show-dialplan.php Sorry about that... Try: http://www.sedwards.com/recover-show-dialplan.txt -- Thanks in advance

Re: [asterisk-users] restart system from extension

2017-04-06 Thread Steve Edwards
he single '/sbin/reboot' command. Do as I say, not as I've done :) -- Thanks in advance, --------- Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve

Re: [asterisk-users] Commit dialplan & other config. in memory to disk?

2017-04-06 Thread Steve Edwards
that through a PHP script that recovered enough. You're welcome to the script at: http://www.sedwards.com/recover-show-dialplan.php -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760

Re: [asterisk-users] Any way to clear ALL gosub stacks without knowing what they are?

2017-04-01 Thread Steve Edwards
On Sat, 1 Apr 2017, Jonathan H wrote: Any way of clearing ALL gosub stacks in dialplan? 1) rm -f /etc/asterisk/extensions.conf? 2) hangup()? (It is April fools...) -- Thanks in advance, - Steve Edwards sedwa

[asterisk-users] SIP reload not changing codecs

2017-02-27 Thread Steve Edwards
9) WriteFormat: slin ReadFormat: g729 Why do I need to restart to get calls to actually use the new codec? -- Thanks in advance, ----- Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST

[asterisk-users] How to show codec packetization?

2017-02-27 Thread Steve Edwards
the the configured and the actual (in use) packetization? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281

Re: [asterisk-users] Turn on SIP debugging from DialPlan

2017-02-17 Thread Steve Edwards
, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new

Re: [asterisk-users] Using g729 now that patents have expired

2017-02-07 Thread Steve Edwards
On Tue, Feb 7, 2017 at 4:47 PM, Steve Edwards <asterisk@sedwards.com> wrote: Now that the g729 patents have expired, how do we use g729 in Asterisk? Will Digium be releasing a g729 codec for 'free' use or do we download the 'free' codec off the Internet now that

[asterisk-users] Using g729 now that patents have expired

2017-02-07 Thread Steve Edwards
, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281 -- _ -- Bandwidth and Colocation Provided by http

Re: [asterisk-users] Call List Campaign to an IVR

2017-02-06 Thread Steve Edwards
to the message when it was most convenient for them. That way, they were in control and things were done on their terms. On 6/02/2017, at 11:34 AM, Steve Edwards <asterisk@sedwards.com> wrote: Love the idea. How? On Mon, 6 Feb 2017, Matt Riddell wrote: exten => _X.,1,Dial(SIP/01

Re: [asterisk-users] Call List Campaign to an IVR

2017-02-06 Thread Steve Edwards
it was most convenient for them. That way, they were in control and things were done on their terms. Love the idea. How? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST

[asterisk-users] SIP host name resolution

2017-02-03 Thread Steve Edwards
in my DNS? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281

Re: [asterisk-users] Callback on busy

2017-01-27 Thread Steve Edwards
nnels. Visit http://www.voip-info.org and search for 'asterisk call back' for examples of how others have approached this problem. -- Thanks in advance, ----- Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.c

Re: [asterisk-users] Kernel/Asterisk/DAHDI/Libpri version matrix?

2017-01-16 Thread Steve Edwards
, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281 -- _ -- Bandwidth and Colocation

Re: [asterisk-users] Kernel/Asterisk/DAHDI/Libpri version matrix?

2017-01-16 Thread Steve Edwards
Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk

[asterisk-users] Kernel/Asterisk/DAHDI/Libpri version matrix?

2017-01-16 Thread Steve Edwards
is the highest version of Asterisk I can run with kernel 2.6.26 and what would be the appropriate versions of DAHDI and libpri? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST

Re: [asterisk-users] RS485 Audio device

2016-11-02 Thread Steve Edwards
On Wed, 2 Nov 2016, Jerry Geis wrote: "AOR" or Area of refuge I have one of those. I call it my 'man cave.' -- Thanks in advance, ----- Steve Edwards sedwa...@sedwards.com Voice: +1-760-46

Re: [asterisk-users] tcpenable

2016-10-19 Thread Steve Edwards
IP address. -- Thanks in advance, ----- Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwar

Re: [asterisk-users] tcpenable

2016-10-19 Thread Steve Edwards
On Wed, 19 Oct 2016, Jerry Geis wrote: I am playing with tcpenable... on 13.11.2 This may yield clues: sudo netstat --all --numeric --program | grep asterisk -- Thanks in advance, - Steve Edwards sedwa

Re: [asterisk-users] Multiple readfile oddities, newlines etc

2016-10-17 Thread Steve Edwards
-- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281

  1   2   3   4   5   6   7   8   9   10   >