Re: [asterisk-users] recreating extensions.conf from live dialplan ?

On Wed, 13 Apr 2016, Steve Edwards wrote: On Wed, 13 Apr 2016, Jeremy Kister wrote: is there a way i can use the asterisk cli (or some other asterisky method) to recreate that extensions.conf ? Will 'dialplan save' help? I just tried this one. It writes the dialplan, but without

Re: [asterisk-users] recreating extensions.conf from live dialplan ?

On Wed, 13 Apr 2016, Jeremy Kister wrote: is there a way i can use the asterisk cli (or some other asterisky method) to recreate that extensions.conf ? Will 'dialplan save' help? -- Thanks in advance, - Steve Edwards

Re: [asterisk-users] implementing asterisk call center.

On 06/04/16 20:58, Goke Aruna wrote: Can someone help me with a kind of howto build call center around asterisk with all the necessary features like CTI, call recordings, call spying, real time monitoring etc? What is your budget? I'm sure there are many contractors who can help. Steve

Re: [asterisk-users] Best timing source?

, - 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 -- New

Re: [asterisk-users] Best timing source?

show us what you have done so far. Most SIP providers have sample snippets for your sip.conf and extensions.conf files. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST

Re: [asterisk-users] Lost outgoing SIP packets

On Thu, 31 Mar 2016, Dovid Bender wrote: Just guessing I would verify that the out of : iptables -L -nv Shows no dropped packets... Doesn't tcpdump 'see' packets before iptables? -- Thanks in advance, - Steve Edwards

Re: [asterisk-users] Is possible to use FXO Digium card like a Fax modem?

have a multi-channel PSTN connected FAX modem. This arrangement is widely used with HylaFAX, although people do use it with other FAX software, such as the stuff built into Windows (using ethernet virtual terminals to connect the windows box to the linux box). Regards, Steve

Re: [asterisk-users] SIP trunk with whatsapp

On 28/03/16 12:46, bilal ghayyad wrote: Does anyone has information if possible to setup SIP trunk with whatsapp? How can we let asterisk send and receive calls from whatsapp? I don't think you can. Whatsapp is a closed system. Steve

Re: [asterisk-users] Incoming INVITE with Portability Info and LRN

On Sun, 20 Mar 2016, Trey Hilyard wrote: On Mar 18, 2016 8:27 PM, "Steve Edwards" <asterisk@sedwards.com> wrote: >> >> On Fri, 18 Mar 2016, Trey Hilyard wrote: >> >>> I thought this would be as easy as >>> exten => _XX\;rn=+1913

Re: [asterisk-users] Incoming INVITE with Portability Info and LRN

On Fri, 18 Mar 2016, Trey Hilyard wrote: I thought this would be as easy as exten => _XX\;rn=+1913663,1,Goto(from_pstn,${EXTEN:0:10}) Have you tried the '_!.' pattern? -- Thanks in advance, ----- Steve Edwa

Re: [asterisk-users] Using Asterisk to play Icecast streams

which may be of questionable legality depending on the stream. -- 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] Incoming INVITE with Portability Info and LRN

On Fri, 18 Mar 2016, Steve Edwards wrote: Have you tried the '_!.' pattern? The '_x.' pattern works fine. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST

Re: [asterisk-users] Incoming INVITE with Portability Info and LRN

"]?set(LRN=${LRN:1})) same = n, goto(${LRN},${DID},1) same = n, hangup() -- Thanks in advance, ----- Steve Edwards sedwa...@sedwards.com V

Re: [asterisk-users] Need help with my dial plan - general logic flaws

to dial operator (0) or 1,2,3 and such. A new context is a great way to limit the scope of trouble after hours callers can get into. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867

Re: [asterisk-users] Dialplan question: Variables in GoTo() ?

template in around 30 contexts and 'goto(${CONTEXT},s,1)' about 15 times. Note that the last example 'nests' the variable expansion -- a variable within a variable. -- Thanks in advance, - Steve Edwards sedwa

Re: [asterisk-users] Asterisk Call Forwarding

-- Thanks in advance, ----- Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- N

Re: [asterisk-users] load test docker images?

, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs

Re: [asterisk-users] Voice recognition IVR Is it possible?

On 22/02/16 23:58, Frank wrote: On Tue, 2016-02-23 at 00:43 +0100, Laszlo wrote: ... Speech API key from Google Yes... OK... but... where and how can I obtain this API Key? Google?... Steve -- _ -- Bandwidth

Re: [asterisk-users] Authenticate() 11.21.0

age to "please enter the password followed by the pound key"? or is there another version of Authenticate() that I'm not aware of or another way to prompt for a password? READ() ? Steve -- _ -- Bandwidth and Colocat

Re: [asterisk-users] How to deal with error messages passed as Early Media

On 03/02/16 15:29, Olivier wrote: 2016-02-03 15:59 GMT+01:00 Steve Howes <steve-li...@geekinter.net <mailto:steve-li...@geekinter.net>>: On 03/02/16 14:41, Olivier wrote: How can I best deal with error messages passed as Early Media. Tell the ITSP to giv

Re: [asterisk-users] How to deal with error messages passed as Early Media

save yourself a whole lot of bother. Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org

Re: [asterisk-users] asterisk-users Digest, Vol 138, Issue 19

conversion to Unix (lf). None of which can be identified with the information provided. The 'bash 5:command not found' snippet implies that something is wrong on the 5th line of your script. -- Thanks in advance, - Steve Edwards

Re: [asterisk-users] how to flush user input before READ()

On Tue, 19 Jan 2016 09:02:30 -0800 (PST) Steve Edwards <asterisk@sedwards.com> wrote: How about a read() to a dummy variable with a 1 second timeout to consume the octothorpe and password? If 1 second is too long, you could write an AGI to use the 'wait for digit' AGI command

Re: [asterisk-users] how to flush user input before READ()

On Mon, 18 Jan 2016, Ethy H. Brito wrote: how to flush user input before READ()? On Mon, 18 Jan 2016 09:38:52 -0800 (PST) Steve Edwards <asterisk@sedwards.com> wrote: How about a read() to a dummy variable with a 1 second timeout to consume the octothorpe and password? On T

Re: [asterisk-users] how to flush user input before READ()

On Mon, 18 Jan 2016, Ethy H. Brito wrote: how to flush user input before READ()? How about a read() to a dummy variable with a 1 second timeout to consume the octothorpe and password? -- Thanks in advance, - Steve

Re: [asterisk-users] Help me please i am facing much trouble

, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every

Re: [asterisk-users] Call Recording

, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live

Re: [asterisk-users] No joy with my first AGI Python script

an application work? 5) Can you play any audio to the channel? Does playback(demo-congrats) work? (That was my last straw to grasp -- need another cup of tea.) -- Thanks in advance,

Re: [asterisk-users] Manipulating of a dialed sequence

e you tried the 'cut' function? -- Thanks in advance, ----- Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST -- _ -- Bandwidth and Colocat

Re: [asterisk-users] SIP calls dropping at 15 minutes

On 11/20/15 11:13 AM, Steve Edwards wrote: I have a problem where SIP calls from some providers are dropping at 15 minutes. The topology is: Client sends calls to a host running OpenSIPS, OpenSIPS sends calls to an Asterisk server. 1) Is a 'ds_select_dst()' followed by a 'forward

[asterisk-users] SIP calls dropping at 15 minutes

y to route calls in OpenSIPS? It works most of the time. 2) Can (or should) I configure Asterisk to not send the INVITE at 15 minutes? 3) Should OpenSIPS be responding differently to the INVITE at 15 minutes? -- Thanks

Re: [asterisk-users] Asterisk Mobile Dialer

On Mon, 2 Nov 2015, Helvio Junior wrote: We are launching a new product to help-us to reduce mobile call costs using Asterisk. Commercial products belong on asterisk-biz. -- Thanks in advance, - Steve Edwards sedwa

Re: [asterisk-users] Is there any API on Free PBX to connect to the extensions?

, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New

Re: [asterisk-users] Remote UNIX connection / disconnected.

can set 'hideconnect = yes' -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST -- _ -- Bandwidth and Colocation

Re: [asterisk-users] Remote UNIX connection / disconnected.

ideas are appreciated. Use AMI instead? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST -- _ -- Bandwidth

Re: [asterisk-users] asterisk core dumped after UBUNTU 14.04 dist-upgrade

s. Different CPU? Different kernel version? Was the install from packages or did you compile it yourself? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com

Re: [asterisk-users] Fwd: Sublime Text License Key

Wonder what happens when an entire mailing list tries to use that key?... On 05/10/15 15:28, Optical Phoenix wrote: -- Forwarded message -- From: *Sublime HQ Pty Ltd* > Date: Wednesday, July 25, 2012 Subject: Sublime Text

Re: [asterisk-users] Fwd: Sublime Text License Key

On 05/10/15 16:18, Mitul Limbani wrote: The company making sublime text gets few thousands of dollars of notional loss :) I was thinking more about if they'd built in software activation type stuff. But yea, stealing bad etc too. Steve

Re: [asterisk-users] pedantic=yes in sip.conf

Alan, A little more context would be useful. Where are you putting the '#' and why? ( If all else fails, print it out and mail it to them ;-) ) %23 is the correct encoding for a hash '#' symbol in many SIP contexts, and should be decoded by a properly functioning far-end. Regards, Steve

Re: [asterisk-users] Real-time payments platform - looking to invite a few users

On Tue, 29 Sep 2015, Shamir Allibhai wrote: Our platform blah, blah, blah. You already pitched on -biz. Asterisk-users is the wrong place to pitch your business. -- Thanks in advance, - Steve Edwards sedwa

Re: [asterisk-users] sox convert gsm file

-- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST -- _ -- Bandwidth and Colocation Provided by http://www.api

Re: [asterisk-users] Fail2ban

to see if he has any new tricks included. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST

Re: [asterisk-users] phones behind nat

and entered 'sudo netstat -a -n -p | grep 5060' -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST -- _ -- Bandwidth

Re: [asterisk-users] Looking for Asterisk Consultants & Experts

On Wed, 2 Sep 2015, Shahid H wrote: Can someone recommend me where is best place to find Asterisk Expert/Consultant for freelance work? Please repost to the asterisk-biz list. -- Thanks in advance, - Steve Edwards

[asterisk-users] AMI 'meetme list concise' hanging

that a second or two? 4) Is there a better 'work-around?' -- Thanks in advance, ----- Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST -- _

Re: [asterisk-users] Anyone doing speech to text?

, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New

Re: [asterisk-users] Anyone doing speech to text?

, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live

Re: [asterisk-users] asterisk server stress test

, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join

Re: [asterisk-users] asterisk server stress test

Please don't top post. On Wed, 19 Aug 2015, James Cass wrote: Steve, would you be willing to share that quick bash script? There's no magic in the script, but here it is, embarrassing myself: cp sample-call-file /tmp/ chmod +x /tmp/sample-call-file for I in $(seq

Re: [asterisk-users] 786 000 files limit Centos 7 - Asterisk keep complaining

, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New

Re: [asterisk-users] AgentRequest() and which agent id?

, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every

Re: [asterisk-users] Asterisk uses Anonymous, but why?

On Thu, 6 Aug 2015, Steve Edwards wrote: Would comparing an INVITE from X-Lite or X-Pro with the INVITE from Asterisk yield any clues? On Thu, 6 Aug 2015, Murthy Gandikota wrote: For Asterisk INVITE please view http://pastebin.com/v15vMax4 For X-Lite INVITE please view http

Re: [asterisk-users] Asterisk uses Anonymous, but why?

comparing an INVITE from X-Lite or X-Pro with the INVITE from Asterisk yield any clues? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST

Re: [asterisk-users] Asterisk uses Anonymous, but why?

On Thu, 6 Aug 2015, Steve Edwards wrote: On Thu, 6 Aug 2015, Steve Edwards wrote: Would comparing an INVITE from X-Lite or X-Pro with the INVITE from Asterisk yield any clues? On Thu, 6 Aug 2015, Murthy Gandikota wrote: For Asterisk INVITE please view http://pastebin.com/v15vMax4 For X

Re: [asterisk-users] Asterisk uses Anonymous, but why?

, but I've always been happy with vitelity.com. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST-- _ -- Bandwidth

Re: [asterisk-users] Call Center

variables. Based on the 'project' description, that's how I would approach it. For specifics, feel free to break out your check book and contact me off-list :) -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com

[asterisk-users] Why no CentOS 7 repos?

Any particular reason CentOS 7 repos aren't available? I'm finding integration issues with CentOS 6's ancient versions of MySQL and PHP with third party applications. -- Thanks in advance, - Steve Edwards sedwa

Re: [asterisk-users] Why no CentOS 7 repos?

On 27/07/2015 1:51 PM, Steve Edwards wrote: Any particular reason CentOS 7 repos aren't available? I'm finding integration issues with CentOS 6's ancient versions of MySQL and PHP with third party applications. On Mon, 27 Jul 2015, Ron Wheeler wrote: You might have o upgrade MySQL and PHP

Re: [asterisk-users] Asterisk 11 and pulseaudio setup as local user

' is because myuser cannot access resources the same way root can -- like an elevated priority or 'real time' or ??? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST

Re: [asterisk-users] Asterisk dialplan best practices syntax

, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided

Re: [asterisk-users] Asterisk dialplan best practices syntax

, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com

Re: [asterisk-users] Asterisk 13 logging to two places

in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000

Re: [asterisk-users] setting outbound caller ID

On Thu, 18 Jun 2015, Matt Riddell wrote: Did you buy the number from your carrier? I prefer using 'rent' instead of 'buy' :) -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867

Re: [asterisk-users] Variable variables

, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth

Re: [asterisk-users] howto copy a voicemail message to another machine ?

in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000

Re: [asterisk-users] small homebrew pbx

so you could only make calls between extensions. 3) For the IP phones, check out ebay.com. Last year, I picked up 3 Polycom SP 501's for $20.00 each. A little dated, but a great phone. -- Thanks in advance, - Steve Edwards

Re: [asterisk-users] small homebrew pbx

On Mon, 15 Jun 2015, Steve Edwards wrote: Although, if you lose power, you've probably lost your Internet connection as well so you could only make calls between extensions. And you would lose the Italian equivalent of 911. In the US, everybody over the age of 6 has a cell phone stapled

Re: [asterisk-users] Calling multiple phones at ones

, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000-- _ -- Bandwidth and Colocation Provided

Re: [asterisk-users] Am I cracked?

, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New

Re: [asterisk-users] Am I cracked?

manageable. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000

Re: [asterisk-users] Am I cracked?

in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000

Re: [asterisk-users] Curious problem with NAT

On Sun, Jun 7, 2015 at 8:46 AM, Steve Totaro stot...@totarotechnologies.com wrote: On Sun, Jun 7, 2015 at 4:49 AM, Luca Bertoncello lucab...@lucabert.de wrote: Ashwin Surendran ashwin.surend...@now-health.com schrieb: What settings have you got for directmedia? Could you try nat

Re: [asterisk-users] Curious problem with NAT

... :( Other idea? Thanks Luca Bertoncello (lucab...@lucabert.de) Are you using the wifi on on the cellphone? The peer IP is showing as 192.168.200.3 which is not a routable address. Unless things have changed, double NAT configurations do not work. Thanks, Steve T

Re: [asterisk-users] Curious problem with NAT

On Sun, Jun 7, 2015 at 10:05 AM, Luca Bertoncello lucab...@lucabert.de wrote: Zitat von Steve Totaro stot...@totarotechnologies.com: Are you using the wifi on on the cellphone? The peer IP is showing as 192.168.200.3 which is not a routable address. Unless things have changed, double NAT

Re: [asterisk-users] Connecting two Asterisk

!!) installed on my Server... Maybe fiddling with the SIP and RTP ports would help. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline

Re: [asterisk-users] Connecting two Asterisk

... -- Call accepted by X.Y.Z.K (format gsm) -- Format for call is gsm I thought GSM regurgitated by cell had issues. Can you try alaw/ulaw? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1

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

of expertise, so if you think I've misconfigured something on my side, please let me know. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline

Re: [asterisk-users] Signaling incoming call

= MYSECRET type= peer 5) I'd try and move more of the common settings to general, but these were the ones listed on voip-info.org. -- Thanks in advance, - Steve Edwards

Re: [asterisk-users] Signaling incoming call

/documentation/howtos/how-to-enable-distinctive-ringing-alert-info-for-calls-from-particular- -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline

Re: [asterisk-users] Debugging dialplan

) [pbx_config] Note the format of my verbose() arguments. It makes it easy to 'cut-n-paste' in a 'dialplan show' command. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867

Re: [asterisk-users] Debugging dialplan

On Fri, 29 May 2015, Steve Edwards wrote: ; admin functions exten = _[456],1, verbose(1,[${EXTEN}@${CONTEXT}!${ANI}]) exten = _[456],n, gotoif($[TRUE = ${ADMIN}]?meetme-star-admin-menu,${EXTEN},1) exten = _[456],n, goto(enter-room

Re: [asterisk-users] chanspy and mixmonitor

On Fri, 29 May 2015, sysad...@reed-media.com wrote: You may get better replies if you start a new thread rather than replying to an unrelated thread. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com

Re: [asterisk-users] Asterisk virtual hosting

On Sun, 17 May 2015, martin f krafft wrote: You know the Henry Ford quote about faster horses, right? ;) I do now :) -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST

Re: [asterisk-users] Asterisk virtual hosting

On Sun, 17 May 2015, martin f krafft wrote: also sprach Steve Edwards asterisk@sedwards.com [2015-05-16 23:22 +0200]: I use a preprocessor (http://software.hixie.ch/utilities/unix/preprocessor/) to tailor dialplans and configuration files to each host based on the client (or project

Re: [asterisk-users] Asterisk virtual hosting

,execif($[${ACCEPT-COUNTER} 0],hangup) #endif -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000

Re: [asterisk-users] Retransmission Timeout results in dropped calls after 32 seconds

rather than RFC compliant SIP. If that is the case Cisco call it a SIP fixup and you just need to disable it. Hope that helps, Steve On Wed, 13 May 2015 at 16:59 Andrew Martin amar...@xes-inc.com wrote: - Original Message - From: Joshua Colp jc...@digium.com To: Asterisk Users Mailing

Re: [asterisk-users] adding area code

On Mon, 27 Apr 2015, Bryant Zimmerman wrote: exten = _9XXX,n,Set(dialnumber=${l_HomeAreaCode}${EXTEN-1}) Missing a colon? ${EXTEN:-1} -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com

Re: [asterisk-users] adding area code

On Mon, 27 Apr 2015, Chad Wallace wrote: On Mon, 27 Apr 2015 14:30:07 -0700 (PDT) Steve Edwards asterisk@sedwards.com wrote: On Mon, 27 Apr 2015, Bryant Zimmerman wrote: exten = _9XXX,n,Set(dialnumber=${l_HomeAreaCode}${EXTEN-1}) Missing a colon? ${EXTEN:-1} Does

Re: [asterisk-users] MixMonitor Files Always Empty

monitor() call path and your mixmonitor() call path? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000

[asterisk-users] meetme vs confbridge max user comparison wanted

server?' -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000

Re: [asterisk-users] Multicast to polycom from asterisk

, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation

[asterisk-users] error retrieving a video voicemail in asterisk 11

Using asterisk 11.16.0 I am unable to retrieve any voicemail with a video attachment while using any video phone. This does work in my 1.8.23.1 installation. The file is skipped with the ast_streamfile error (and moved to OLD), and the prompts following that message display the ast_best_codec

Re: [asterisk-users] exten versus EXTEN

as '=' when used in the dialplan. Personally, I always use '=' as the dialplan doesn't seem to be the place for some object oriented mumbo-jumbo -- at least to my 'C programmer till I die' eyes. -- Thanks in advance, - Steve

Re: [asterisk-users] [OT] switches

, but I don't remember if it was device or server discovery. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731

Re: [asterisk-users] Asterisk 13 : SILK codec ?

. Are these codecs no longer going to be available for 13 and up? Or, were they just overlooked in the day-to-day rush called life? murf​ -- Steve Murphy ParseTree Corporation -- _ -- Bandwidth and Colocation Provided by http://www.api

Re: [asterisk-users] Regarding Text To Speech conversion

, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New

Re: [asterisk-users] hangup call gw FXO

Looking at the pastebin, the Vega device sends a CANCEL with reason: Reason: Q.850 ;cause=16. Cause 16 is normal clearing and suggests that the original caller has disconnected. I would take a look at the Vega's logs Regards, Steve On Thu, 5 Mar 2015 at 11:41 ricky gutierrez xserverli

Re: [asterisk-users] Problems with the voice quality under load

idle. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000

Re: [asterisk-users] System() command refuses to execute bash script

, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided

Re: [asterisk-users] System() command refuses to execute bash script

=${PATH}\ $ASTERISK $START_OPTIONS -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1

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