Re: [asterisk-users] PJSIP to Twilio over TLS - wildcard cert problem

fan on wildcards. then le came along, and then added dns01 support. now i prefer a separate cert each plus a 3/1/1 tlsa for each port. but at the time it was anoying. -JimC -- James Cloos OpenPGP: 0x997A9F17ED7DAEA6 -- _

Re: [asterisk-users] PJSIP to Twilio over TLS - wildcard cert problem

advise as set in stone, and so asterisk refuses such certs. i doubt that stance is different under sangoma. the only workaround is to remind twil of the rfc and get them to replace the wildcard with an rfc-copliant cert. at least for the sip ports. -JimC -- James Cloos

[asterisk-users] ?????? Does Asterisk-16.1.1 support "make freepbx"

16.1.1 support "make freepbx" On Sun, 30 Dec 2018 at 15:10, Joshua C. Colp wrote: > > On Sun, Dec 30, 2018, at 9:03 AM, james wrote: > > hello: > > Does asterisk-16.1.1 support freepbx by default? > > No version of Asterisk currently has any built in mechanism to i

[asterisk-users] Does Asterisk-16.1.1 support "make freepbx"

hello: Does asterisk-16.1.1 support freepbx by default? /contrib/scripts/install_prereq install ./configure make make install make config make freepbx // I run this command without success. after then, we can access the freepbx GUI? this website show it work:

Re: [asterisk-users] T-38 re-invite issue

>>>>> D'Arcy Cain writes: >> Ie after both sides select t38, until they agree on the t38 terms. > OK, so does that mean that setting it to 25000 should leave time for the > re-invite or does the timeout start after that. As I wrote above, after that. After the sip/

Re: [asterisk-users] T-38 re-invite issue

ce it starts. Ie after both sides select t38, until they agree on the t38 terms. -JimC -- James Cloos OpenPGP: 0x997A9F17ED7DAEA6 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the

Re: [asterisk-users] OT: Explain where mailing list bouncing comes from ?

sent by e-Mail James B. Byrnemailto:byrn...@harte-lyne.ca Harte & Lyne Limited http://www.harte-lyne.ca 9 Brockley Drive vox: +1 905 561 1241 Hamilton, Ontario fax: +1 905 561 0757 Canada L8E

[asterisk-users] Snom870 FW:8.7.5.35

James B. Byrnemailto:byrn...@harte-lyne.ca Harte & Lyne Limited http://www.harte-lyne.ca 9 Brockley Drive vox: +1 905 561 1241 Hamilton, Ontario fax: +1 905 561 0757 Canada L8E

Re: [asterisk-users] OT: DMARC enabled domains on this list

Do NOT transmit sensitive data via e-Mail Do NOT open attachments nor follow links sent by e-Mail James B. Byrnemailto:byrn...@harte-lyne.ca Harte & Lyne Limited http://www.harte-lyne.ca 9 Brockley Drive vox: +1 905 561 1241 Hamilton, Ontario

Re: [asterisk-users] OT: DMARC enabled domains on this list

-Mail James B. Byrnemailto:byrn...@harte-lyne.ca Harte & Lyne Limited http://www.harte-lyne.ca 9 Brockley Drive vox: +1 905 561 1241 Hamilton, Ontario fax: +1 905 561 0757 Canada L8E

Re: [asterisk-users] Let's encrypt privkey : Specified certificate file could not be used

sterisk, run this as root: su -c 'cat /etc/letsencrypt/live/ws.mydomain.tld/privkey.pem' - asterisk If it fails, then the problem is permissions. You may need to alter the permissions on /etc/letsencrypt to allow non-root uids to access the symlinks and their targets. -JimC -- James Cloos <cl...@jh

[asterisk-users] SNOM870 provisioning BLF settings

Do NOT transmit sensitive data via e-Mail Do NOT open attachments nor follow links sent by e-Mail James B. Byrnemailto:byrn...@harte-lyne.ca Harte & Lyne Limited http://www.harte-lyne.ca 9 Brockley Drive vox: +1 905 561 1241 Hamilton, Onta

Re: [asterisk-users] iaxModem pickup problem

tive data via e-Mail Do NOT open attachments nor follow links sent by e-Mail James B. Byrnemailto:byrn...@harte-lyne.ca Harte & Lyne Limited http://www.harte-lyne.ca 9 Brockley Drive vox: +1 905 561 1241 Hamilton, Ontario fax: +1 905 561 0

Re: [asterisk-users] iaxModem pickup problem

e-Mail is NOT a SECURE channel *** Do NOT transmit sensitive data via e-Mail Do NOT open attachments nor follow links sent by e-Mail James B. Byrnemailto:byrn...@harte-lyne.ca Harte & Lyne Limited http://www.harte-lyne.ca 9 Brockley Drive vox: +1 905 56

Re: [asterisk-users] iaxModem pickup problem

On Thu, May 4, 2017 10:22, James B. Byrne wrote: I am advised that it may be possible thast the astdb.sqlite3 database may be corrupted. Are there procedures to rebuild or repair this? Where are they documented? If not then how does one repair such? -- *** e-Mail is NOT a SECURE

[asterisk-users] iaxModem pickup problem

. -- *** e-Mail is NOT a SECURE channel *** Do NOT transmit sensitive data via e-Mail Do NOT open attachments nor follow links sent by e-Mail James B. Byrnemailto:byrn...@harte-lyne.ca Harte & Lyne Limited http://www.harte-lyne.ca 9 Brockley D

Re: [asterisk-users] log incoming calls without answering

o->sip gateway will cancel the sip call just like it would if the caller hung up. (There is a possibility that any given gateway may not cancel the sip call until the analog call is completed; you need to test.) -JimC -- James Cloos <

Re: [asterisk-users] Voicemail asking for login

exec()), including a log at the start of what is in *data and args. Looking at it, it only plays vm-whichbox when ast_strlen_zero(data), which implies that the args to Voicemail are not making it through. -JimC -- James Cloos <cl...@jhcloos.

Re: [asterisk-users] Voicemail asking for login

I enable full log and run 'core set debug 9' before doing a pair of tests. (The full log is easier to grep than the console output.) Then compare a working vs stocktrans2 side by side. -JimC -- James Cloos <cl...@jhcloos.com> OpenPGP: 0x997A9F17ED

Re: [asterisk-users] E-911

in the SIP From: header.) -JimC -- James Cloos <cl...@jhcloos.com> OpenPGP: 0x997A9F17ED7DAEA6 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community

Re: [asterisk-users] inbound T38 to email

applications. They can be configured (in res_fax.conf) to use t38 when available. -JimC -- James Cloos <cl...@jhcloos.com> OpenPGP: 0x997A9F17ED7DAEA6 -- _ -- Bandwidth and Colocation Provided by http://www.api

Re: [asterisk-users] Problem with REMAINDER? 957%60 be 15 remainder 57 not 15 remainder -3 ?

the rounding mode), fmod(3) is defined to trunc(3)ate the quotient. So the result of x%y will always be in the range [0,x] and the results of remainder(x,y) will be in the range (-y/2,y/2]. -JimC -- James Cloos <cl...@jhcloos.com> OpenPGP: 0x997A9F17ED

Re: [asterisk-users] Problem with REMAINDER? 957%60 be 15 remainder 57 not 15 remainder -3 ?

All I can tell you is where -3 comes from. >From http://www.voip-info.org/wiki/view/Asterisk+Expressions : REMAINDER(x,y) computes the remainder of dividing x by y. The return value is x - n*y, where n is the value x/y, rounded to the nearest integer. If this quotient is 1/2, it is rounded to the

Re: [asterisk-users] pbx testsuite

I'd like to see it. That definitely interests me. Thanks Glen On Jun 15, 2016 9:05 AM, "Marek Červenka" wrote: > hi, > > we have in house developed pbx testsuite based on > >- node.js >- selenium >- protractor >- gulp >- pjsip - pjsua python >- docker

Re: [asterisk-users] Windstream SIP Trunk settings

insecure=port,invite Incoming settings - none Registration string - username:passw...@xxx.xxx.xxx.xxx James Cass <http://goog_987864563> jcas...@gmail.com On Tue, Feb 23, 2016 at 12:20 PM, Rodrigo Ramírez Norambuena < decipher...@gmail.com> wrote: > February 23 2016 9:37 AM, &qu

Re: [asterisk-users] Windstream SIP Trunk settings

Thanks everyone, all sound advice. Still can't even get the calls to show up on the console at all - I suspect the issue is on the WS side, as I'm not having any issues with other carriers with similar settings. Thanks again. James Cass <http://goog_987864563> jcas...@gmail.com On Mon,

[asterisk-users] Windstream SIP Trunk settings

Does anyone on this list use Windstream as a SIP trunk provider? If so, would you mind sharing your peer settings? I'm using asterisk 13.7.2 and can't seem to get the inbound working correctly (using registration). Outbound is fine, but they are seeing an authentication error on their end.

[asterisk-users] Asterisk & Docker

<==> 192.168.50.1 <==> 192.168.50.254 <==> External Internet IP It seems to me that we are doing double NAT? -- James McDonald IT Services 11/79 Earl St, Kew, VIC, 3101 Mob.: +61 428 964 633 Email: ja

Re: [asterisk-users] PJSIP Stun/ICE

n some places (including here) static ip is not affordable. -JimC -- James Cloos <cl...@jhcloos.com> OpenPGP: 0x997A9F17ED7DAEA6 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Ast

Re: [asterisk-users] PJSIP Stun/ICE

at announce you'd have received -- I expect -- quite a few complaints. This flies in the face of all of the (very welcome) work which went into supporting reload rather than restart. Getting pjsip to support changes on a reload would be an acceptable first step. -JimC -- James Clo

Re: [asterisk-users] PJSIP Stun/ICE

ck of full support for traversing nat makes pjsip worthless for a large number of users. And the whole point of realtime is to have all of the rt config fully dymanic. If ari cannot avoid that limitation, chan_sip should get full ongoing maintainance until pjsip is fixed. -JimC -- Jame

[asterisk-users] ARI all subscribe

to use wscat with such a sub to get a better idea of what the various events look like. Thanks, -JimC -- James Cloos <cl...@jhcloos.com> OpenPGP: 0x997A9F17ED7DAEA6 -- _ -- Bandwidth and Colocation Provided by http://w

Re: [asterisk-users] asterisk server stress test

Steve, would you be willing to share that quick bash script? James Cass http://goog_987864563 jcas...@gmail.com On Wed, Aug 19, 2015 at 12:11 PM, Steve Edwards asterisk@sedwards.com wrote: On Wed, 19 Aug 2015, Dominique Haeber wrote: Hi Barry Flanagan, Barry Flanagan barryf-li

Re: [asterisk-users] How to enable IM over the asterisk server

You can have the openfire server installed on the same server as asterisk without any issue, just size your server appropriately. Just keep in mind they are different services. James Cass http://goog_987864563 jcas...@gmail.com On Wed, Jul 8, 2015 at 4:24 AM, Thyda ENG ength...@gmail.com wrote

Re: [asterisk-users] How to enable IM over the asterisk server

The asterisk plugin for openfire would be what I would think would do that, but as another person posted, it's very deprecated, so I'm not sure how well it would work. I've never used it personally. James Cass http://goog_987864563 jcas...@gmail.com On Wed, Jul 8, 2015 at 11:45 AM, Thyda ENG

Re: [asterisk-users] small homebrew pbx

I picked up a cheap JS200-FX on ebay: http://x100p.com/products/js200fx.php for $30, and it works great for a home install. Very low power draw as well. James Cass http://goog_987864563 jcas...@gmail.com On Mon, Jun 15, 2015 at 10:50 AM, Kevin Larsen kevin.lar...@pioneerballoon.com wrote

Re: [asterisk-users] sedwa...@sedwards.com causes me to be knocked off the list

I agree with Doug, everything looks legit for sedwards.com from my point of view. Giles, maybe there's something wrong with the DNS server you're using? James Cass http://goog_987864563 jcas...@gmail.com On Wed, Jun 3, 2015 at 11:01 AM, Giles Coochey gi...@coochey.net wrote: Someone

Re: [asterisk-users] Phones don't stop ringing when queue is answered

A few things I would try- Change WaitExten to Wait(2) Change Queue(queue_level_1,rtnC,18) to Queue(queue_level_1,rtnC,,,18) Add an Answer() after the first Wait(2) -- _ -- Bandwidth and Colocation Provided by

Re: [asterisk-users] Phones don't stop ringing when queue is answered

What purpose do the WaitExten()s serve here? Are you really allowing the caller to connect to different extensions in the test-queue context? Have you tried without the WaitExten()s? -- _ -- Bandwidth and Colocation Provided by

Re: [asterisk-users] I'm not able to register Softphone(X-lite) in asterisk(Which is installed in EC2 Cloud).

open to manage the server, but you already have that. That's all you need from a security group perspective. Make sure that right then we can inspect your config files. On Apr 28, 2015 1:09 AM, akhilesh chand omakhileshch...@gmail.com wrote: Hi James, Please let me know how could I implement

Re: [asterisk-users] I'm not able to register Softphone(X-lite) in asterisk(Which is installed in EC2 Cloud).

Akhilesh, I have implemented several ec2 instances with both sip and iax2 and have no problems with xlite or hard phones. Have you already opened the ports in the vpc security group on the Amazon side? Let me know is I can help. --James On Apr 24, 2015 3:34 AM, akhilesh chand omakhileshch

Re: [asterisk-users] Script to Program Snom Phones

files therein with names following the specific nomenclature employed buy the phones themselves. Finally you must also set the phones to read from that location and to apply the configurations retrieved therefrom. -- *** E-Mail is NOT a SECURE channel *** James B. Byrne

Re: [asterisk-users] Anonymous SIP calls

may also want to look into getting an ISN number, check out http://freenum.org/ for the details. -JimC -- James Cloos cl...@jhcloos.com OpenPGP: 0x997A9F17ED7DAEA6 -- _ -- Bandwidth and Colocation Provided by http

Re: [asterisk-users] Anonymous SIP calls

-Mail is NOT a SECURE channel *** James B. Byrnemailto:byrn...@harte-lyne.ca Harte Lyne Limited http://www.harte-lyne.ca 9 Brockley Drive vox: +1 905 561 1241 Hamilton, Ontario fax: +1 905 561 0757 Canada L8E 3C3

[asterisk-users] Anonymous SIP calls

call should be handled from a theoretical point of view. Any guidance would be welcome. -- *** E-Mail is NOT a SECURE channel *** James B. Byrnemailto:byrn...@harte-lyne.ca Harte Lyne Limited http://www.harte-lyne.ca 9 Brockley Drive vox: +1

[asterisk-users] Jitsi, SRTP and Asterisk 11

Does anyone here have a Jitsi softphone set up with Asterisk such that SRTP is enabled, TLS is used to pass the SRTP key, and it works? Anyone? If so then what are the settings required for Asterisk? -- *** E-Mail is NOT a SECURE channel *** James B. Byrne

[asterisk-users] Guidence in DialPlan programming.

return to the original referring context ([from-internal-xfer]). -- *** E-Mail is NOT a SECURE channel *** James B. Byrnemailto:byrn...@harte-lyne.ca Harte Lyne Limited http://www.harte-lyne.ca 9 Brockley Drive vox: +1 905 561 1241 Hamilton

Re: [asterisk-users] OT - How does the blind transfer function work on Snom-870?

channel *** James B. Byrnemailto:byrn...@harte-lyne.ca Harte Lyne Limited http://www.harte-lyne.ca 9 Brockley Drive vox: +1 905 561 1241 Hamilton, Ontario fax: +1 905 561 0757 Canada L8E 3C3

Re: [asterisk-users] OT - How does the blind transfer function work on Snom-870?

On Thu, March 5, 2015 05:30, Ruben Rögels wrote: Am 05.03.2015 um 01:09 schrieb James B. Byrne: I am trying to determine how the transfer button on the Snom-870 works with Asterisk. Is the ## special code employed as when it is entered through the handset or is the blind transfer through

[asterisk-users] OT - How does the blind transfer function work on Snom-870?

channel *** James B. Byrnemailto:byrn...@harte-lyne.ca Harte Lyne Limited http://www.harte-lyne.ca 9 Brockley Drive vox: +1 905 561 1241 Hamilton, Ontario fax: +1 905 561 0757 Canada L8E 3C3

Re: [asterisk-users] TLS, SRTP, Asterisk11 and Snom870s

.internal.hamilton.harte-lyne.ca:5061 SIP/2.0 Via: SIP/2.0/UDP 192.168.6.112:5060;branch=z9hG4bK-udx92poqese6;rport From: James B Byrne sip:41...@voinet09.internal.hamilton.harte-lyne.ca:5061;tag=frgaimnglp To: James B Byrne sip:41...@voinet09.internal.hamilton.harte-lyne.ca:5061 Call-ID: 71004941-gk6y4evf6dci

[asterisk-users] TLS, SRTP, Asterisk11 and Snom870s

callerid=James B Byrne 41712 callcounter=yes faxdetect=no cc_monitor_policy=generic If I change the transport setting to TLS then I get this reported: [2015-03-03 11:10:08] ERROR[22244]: tcptls.c:875 ast_tcptls_client_start: Unable to connect SIP socket to 192.168.6.112:5060: Connection refused

Re: [asterisk-users] TLS, SRTP, Asterisk11 and Snom870s

JBB == James B Byrne byrn...@harte-lyne.ca writes: JBB tcpenable=yes JBB tlsenable=yes JBB tlscertfile=/etc/pki/asterisk/ca.harte-lyne.hamilton.asterisk.crt JBB tlscafile=/etc/pki/tls/certs/ca-bundle.crt JBB tlsdontverifyserver=yes JBB tlscipher=ALL JBB tlsclientmethod=tlsv1 You are missing

Re: [asterisk-users] TLS, SRTP, Asterisk11 and Snom870s

MOH Suggest: Voice Mail Extension: *97 -- *** E-Mail is NOT a SECURE channel *** James B. Byrnemailto:byrn...@harte-lyne.ca Harte Lyne Limited http://www.harte-lyne.ca 9 Brockley Drive vox: +1 905 561 1241 Hamilton, Ontario

Re: [asterisk-users] TLS, SRTP, Asterisk11 and Snom870s

are communicating with each other and have been for the past two years. The Snoms are configured for AES-80 and SRTP is enabled on the FreePBX device entry. We have a working PBX system. I am trying to secure it. -- *** E-Mail is NOT a SECURE channel *** James B. Byrne

Re: [asterisk-users] TLS, SRTP, Asterisk11 and Snom870s

like this: register = tls://username:xxx...@sip-tls-proxy.example.org (copied from the example sip.conf). Set tlsbindaddr to the address to which to bind(2) the tls socket. tlsbindaddr=0.0.0.0 is typical in ipv4-only configs. -JimC -- James Cloos cl...@jhcloos.com OpenPGP

Re: [asterisk-users] TLS, SRTP, Asterisk11 and Snom870s

channel *** James B. Byrnemailto:byrn...@harte-lyne.ca Harte Lyne Limited http://www.harte-lyne.ca 9 Brockley Drive vox: +1 905 561 1241 Hamilton, Ontario fax: +1 905 561 0757 Canada L8E 3C3

Re: [asterisk-users] TLS, SRTP, Asterisk11 and Snom870s

On Tue, March 3, 2015 16:34, James Cloos wrote: Other things to consider: The transport config, which can be in [general] or in a peer's [] block. if you want tls-only, use transport=tls it also accepts tcp, udp or a comma-separated list. if given a list, it tries them in order

Re: [asterisk-users] TLS, SRTP, Asterisk11 and Snom870s

On Tue, March 3, 2015 13:37, James Cloos wrote: JBB == James B Byrne byrn...@harte-lyne.ca writes: JBB tcpenable=yes JBB tlsenable=yes JBB tlscertfile=/etc/pki/asterisk/ca.harte-lyne.hamilton.asterisk.crt JBB tlscafile=/etc/pki/tls/certs/ca-bundle.crt JBB tlsdontverifyserver=yes JBB

Re: [asterisk-users] Weird SIP stuff

. Those show the path of the SIP. In your example, look for a Via which mentions 65.211.180.237. Note that the From does not necessary have to be a sip address reachable from the outside. -JimC -- James Cloos cl...@jhcloos.com OpenPGP: 0x997A9F17ED7DAEA6

Re: [asterisk-users] High resident memory with 11.14.0 ?

On Wed, Nov 26, 2014 at 3:20 PM, James Lamanna jlama...@gmail.com wrote: On Tue, Nov 25, 2014 at 10:21 AM, James Lamanna jlama...@gmail.com wrote: On Tue, Nov 25, 2014 at 8:14 AM, Matthew Jordan mjor...@digium.com wrote: On Mon, Nov 24, 2014 at 2:12 PM, James Lamanna jlama...@gmail.com

Re: [asterisk-users] High resident memory with 11.14.0 ?

On Tue, Nov 25, 2014 at 10:21 AM, James Lamanna jlama...@gmail.com wrote: On Tue, Nov 25, 2014 at 8:14 AM, Matthew Jordan mjor...@digium.com wrote: On Mon, Nov 24, 2014 at 2:12 PM, James Lamanna jlama...@gmail.com wrote: Also, how big does the cache in frame.c grow to? I've recompiled

Re: [asterisk-users] High resident memory with 11.14.0 ?

On Tue, Nov 25, 2014 at 8:14 AM, Matthew Jordan mjor...@digium.com wrote: On Mon, Nov 24, 2014 at 2:12 PM, James Lamanna jlama...@gmail.com wrote: Also, how big does the cache in frame.c grow to? I've recompiled with MALLOC_DEBUG on that server: asterisk -rx memory show summary

Re: [asterisk-users] High resident memory with 11.14.0 ?

of that. I can't see what hardware you are using but I think you need to check that the rule above fits your hardware. b.r. Freddi On Fri, Nov 21, 2014 at 10:53 AM, James Lamanna jlama...@gmail.com wrote: Hi, I have an Asterisk server that's been running now for around 2 days. I've

Re: [asterisk-users] High resident memory with 11.14.0 ?

Also, how big does the cache in frame.c grow to? I've recompiled with MALLOC_DEBUG on that server: asterisk -rx memory show summary 1780466242 bytes (1780181594 cache) in2352909 allocations in file frame.c ... Seems like a ridiculous cache. On Mon, Nov 24, 2014 at 9:02 AM, James

[asterisk-users] Size of frame.c cache in Asterisk 11?

frame.c ~$ asterisk -rx core show uptime System uptime: 2 days, 11 hours, 12 minutes, 12 seconds Last reload: 2 days, 11 hours, 12 minutes, 12 seconds $ asterisk -rx core show channels 34 active channels 17 active calls 13824 calls processed This seems like very odd behavior. Thanks. -- James

[asterisk-users] Dahdi fxo vs sip blf

far have failed. At the moment I'm on 12.7, but still using chan_sip. Converting the chan_pjsip will be the next project for this box. What is the proper way to set this up? -JimC -- James Cloos cl...@jhcloos.com OpenPGP: 0x997A9F17ED7DAEA6

[asterisk-users] High resident memory with 11.14.0 ?

active calls? I am not using any realtime peers. There are 100 registered SIP peers on this server as well. Thanks. -- James -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us

Re: [asterisk-users] High resident memory with 11.14.0 ?

Its up to 5.8G of resident memory with 28321 calls processed. The OOM killer is going to kill this soon at this rate (8GB RAM machine). This seems like a pretty serious problem. It looks like I'll need to restart asterisk every night On Fri, Nov 21, 2014 at 10:53 AM, James Lamanna jlama

Re: [asterisk-users] Setting channel musicclass from AGI

Hi Matt, So this actually works (haven't had a chance to try it)? SET VARIABLE CHANNEL(musicclass) default Because musicclass is piece of channel information. Referencing ${musicclass} is not the same thing. Thanks. -- James On Sun, Oct 5, 2014 at 8:05 PM, Matthew Jordan mjor...@digium.com

[asterisk-users] Setting channel musicclass from AGI

Hi, Since SetMusicOnHold() is being deprecated, how do we set the channel musicclass from an AGI script? Last time I checked you can't call dialplan functions from AGI. Thanks. -- James -- _ -- Bandwidth and Colocation Provided

Re: [asterisk-users] Show Log(NOTICE) messages on the console

logger.conf... You should start by comparing that file between the two servers. Not sure if it's still called logger.conf in asterisk 11 though. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to

Re: [asterisk-users] if statement recording - after hours

GotoIfTime() Check out- http://www.voip-info.org/wiki/view/Asterisk+cmd+GotoIfTime If the time is within a certain range, execute the recording dialplan. If it's outside the range, then skip to the dialplan after the recording stuff. --

Re: [asterisk-users] Asterisk-eSpeak and Asterisk 12

that. The option space for espeak has large variability. Flite also needs such tuning for nice output. -JimC -- James Cloos cl...@jhcloos.com OpenPGP: 0x997A9F17ED7DAEA6 -- _ -- Bandwidth and Colocation Provided

Re: [asterisk-users] SIP trunk gives fuzzy / distorted audio on mobiles, OK on fixed lines

Is the quality the same incoming from mobile as outgoing to mobile? On Wed, Jul 30, 2014 at 4:51 AM, A J Stiles asterisk_l...@earthshod.co.uk wrote: I'm having a problem with a new SIP trunk. Calls within the UK to fixed lines are fine, but calls to mobiles have noticeably poorer audio

Re: [asterisk-users] Interesting new hack attack

On 5/22/2014 12:41 PM, Steve Murphy wrote: So, these defenses can be employed to stop/ameliorate such hacking efforts: 1. Keep your phones behind a firewall. Travellers, beware! Never leave the default login info of the phone at default! 2. Never use the default provisioning URL for the

[asterisk-users] FollowMe reinvites

for both the incoming and outgoing legs, it is a bit of a waste to proxy the rtp. And even when the legs are associated with different remotes, I'd prefer to proxy only when NATs a/or v4-v6 gatewaying are involved. -JimC -- James Cloos cl...@jhcloos.com OpenPGP: 0x997A9F17ED7DAEA6

Re: [asterisk-users] new install: no re-invite and unwanted transcoding

I can't help on the first issue, but for the second have you tried doing Set(${SIP_CODEC}=ulaw) before dialing the trunk? I'm in a similar situation where we have g722 internally but our trunk provider only offers ulaw so I see g722-slin-ulaw transcoding. I'm thinking of trying it here (on

Re: [asterisk-users] CBAnn channel not going away in Asterisk 12

On 5/1/2014 10:38 AM, Richard Kenner wrote: Please go ahead and open an issue and attach the refs log and the full DEBUG log. That will allow us to understand what's occurring here. I need to wait until I'm sure this isn't something I caused somehow, so I need to first understand why I'm

Re: [asterisk-users] Proper way to make Asterisk recognize SIP trunk of incoming INVITE when IP is not available

-- James Cloos cl...@jhcloos.com OpenPGP: 0x997A9F17ED7DAEA6 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http

Re: [asterisk-users] srtp/dtls when sip is clear over lo

. :) JColp Yes. Thanks! -JimC -- James Cloos cl...@jhcloos.com OpenPGP: 0x997A9F17ED7DAEA6 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar

Re: [asterisk-users] srtp/dtls when sip is clear over lo

on the matter of what other endpoints are willing to do in such cases? Thanks, -JimC -- James Cloos cl...@jhcloos.com OpenPGP: 0x997A9F17ED7DAEA6 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New

Re: [asterisk-users] Proper way to make Asterisk recognize SIP trunk of incoming INVITE when IP is not available

, but nothing quite worked. -JimC -- James Cloos cl...@jhcloos.com OpenPGP: 0x997A9F17ED7DAEA6 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory

[asterisk-users] srtp/dtls when sip is clear over lo

, but will doing so also block secure media? -JimC -- James Cloos cl...@jhcloos.com OpenPGP: 0x997A9F17ED7DAEA6 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live

Re: [asterisk-users] Open Source Asterisk Polling Solution

them out based on the number of concurrent outbound calls your provider will allow. Hello James, Good to see you here, and thank you very much. Though my basic idea of how it will look using call files and dialplan is like what you and others on here have pointed out. Yes, we are using SIP

Re: [asterisk-users] Open Source Asterisk Polling Solution

On 4/22/2014 5:54 PM, Nick Cameo wrote: Hello Everyone, Thank you all for your response. The people I am doing it for run a non-profit charity, and are legally able to reach out to their customers. I will wire it up to the DNC however, for starters, I would like to get asterisk to: i) Iterate

Re: [asterisk-users] Open Source Asterisk Polling Solution

On 4/21/2014 1:47 PM, Mitul Limbani wrote: Use vicidial for achieving the same. Or call files (or AMI originate), a short bit of dialplan logic, and maybe a call to Queue(). -- _ -- Bandwidth and Colocation Provided by

Re: [asterisk-users] Open Source Asterisk Polling Solution

On 4/21/2014 3:58 PM, Nick Cameo wrote: On Mon, Apr 21, 2014 at 2:01 PM, James Sharp ja...@fivecats.org mailto:ja...@fivecats.org wrote: On 4/21/2014 1:47 PM, Mitul Limbani wrote: Use vicidial for achieving the same. Or call files (or AMI originate), a short bit of dialplan

Re: [asterisk-users] IAXModem or T38Modem?

for things like sub-account, peer and/or trunk configs.) -JimC -- James Cloos cl...@jhcloos.com OpenPGP: 1024D/ED7DAEA6 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us

Re: [asterisk-users] Numbers hackers call

On 3/26/2014 12:20 PM, Michelle Dupuis wrote: If this is to 972 area code then the next digits should be 0X or 0XX but they are not. This differs from what I found documented for that area code - I thought someone from the region might add to the discussion. Not sure if this reflected a

Re: [asterisk-users] IAXModem or T38Modem?

reliable so far as I am aware and I would be made aware pretty quickly if it was not. -- *** E-Mail is NOT a SECURE channel *** James B. Byrnemailto:byrn...@harte-lyne.ca Harte Lyne Limited http://www.harte-lyne.ca 9 Brockley Drive vox: +1 905

[asterisk-users] Which is more efficient for 1 to many broadcasting?

Putting a whole bunch of people into a listen-only/muted Confbridge conference or getting the broadcaster audio into a MOH class and then just having callers attach to that MOH class? Does the the muted side of a Confbridge Room still try to mix in audio from the muted channels or does it

Re: [asterisk-users] Which is more efficient for 1 to many broadcasting?

On 3/18/2014 6:58 PM, Paul Belanger wrote: On Tue, Mar 18, 2014 at 1:02 PM, James Sharp ja...@fivecats.org wrote: Putting a whole bunch of people into a listen-only/muted Confbridge conference or getting the broadcaster audio into a MOH class and then just having callers attach to that MOH

Re: [asterisk-users] Replying to Posts

in the rules. And do not even start on the Chevy vs. Ford debate respecting the technical superiority of Pine over Outlook. GAWD... Life its too short as it is. -- *** E-Mail is NOT a SECURE channel *** James B. Byrnemailto:byrn...@harte-lyne.ca Harte Lyne Limited

Re: [asterisk-users] Host = Dynamic in a Register Free Setup

On 2/18/2014 2:09 PM, Eric Wieling wrote: No. Asterisk will accept calls from unregistered devices, but you have to enable guests I sip.conf and hope your dialplan is secure. No sane person does this. Asterisk cannot send calls to a device unless it knows the address from a register or

[asterisk-users] Looking for some guidance with the Asterisk 12 ARI/API

Hi - I figured this was probably the best place to ask this question Is there anyone that has done anything practical with the API and/or Real Time Database config? If so, I would like to pick your brains if I may. Thanks - G --

Re: [asterisk-users] answering machine screening with MixMonitor

On 2/5/2014 12:09 PM, G. Paul Ziemba wrote: I'm using asterisk 1.8 as an answering machine. I'd like to hear the calls it answers aloud in case I want to pick up and interrupt the call. There are a few articles describing, for example, three-way calling a monitor phone set to auto-answer, but I

[asterisk-users] Asterisk 11 - looking for ideas and a possible solution

First, let me thank all of you for your input - I was looking for some sort of an API to interface with Asterisk via REST. Python, Ruby or something that we could webify using PHP. Some of you suggested Asterisk 12 - I love the idea, but unfortunately, we have an Asterisk 11 install. It seems,

Re: [asterisk-users] Asterisk 11 - looking for ideas and a possible solution

...@digium.com wrote: On 14-01-25 09:54 AM, James Wystead wrote: First, let me thank all of you for your input - I was looking for some sort of an API to interface with Asterisk via REST. Python, Ruby or something that we could webify using PHP. It all depends on what you want to do the with API

Re: [asterisk-users] Asterisk 11 - looking for ideas and a possible solution

understand that there is a real time database module. Thanks - G On Sat, Jan 25, 2014 at 9:47 AM, Joshua Colp jc...@digium.com wrote: On 14-01-25 10:44 AM, James Wystead wrote: Oh - no kidding! What we want to do is be able to create users, voicemail accounts and some of the basic features

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