[asterisk-users] SIP option for Google Fi or Ting.com or similar - 911 is difficult.

​Want to integrate my cellular service into my asterisk dial plan, so this requires SIP capabilities such as one can get with Vitelity vMobile. Does Google Project Fi or Ting or another offer a SIP integration option, but not advertise it? A drawback with Vitelity vMobile is that it will not

[asterisk-users] Favorite this issue to Vote to enable Google Chromium to natively play WAV type 49 or 0x0031 (an asterisk default voice mail format).

Issue 375978 :Support AGSM audio codec https://code.google.com/p/chromium/issues/detail?id=375978#c12 Please click on the link and at least Star / Favorite the issue to vote it up. Chrome can play many audio files directly within the

[asterisk-users] wav49 VoiceMails should play natively in Google Chrome HTML5 - bug report

Please favorite the following bug and express desire to have wav49 VoiceMails play natively in Google Desktop Chrome HTML5. https://code.google.com/p/chromium/issues/detail?id=465431 .WAV already plays in Android Chrome and Chromebook and others, but not the desktop version. --

Re: [asterisk-users] CA Issued Certificates / TLS + SRTP

- External Feature Requests https://jira.polycom.com:8443/browse/EXT - EXT-4669 https://jira.polycom.com:8443/browse/EXT-4669 Please add StartSSL.com StartCOM Certificate Authorityhttps://jira.polycom.com:8443/browse/EXT-4669 https://jira.polycom.com:8443/browse/EXT-4669 On Wed, Feb 1,

Re: [asterisk-users] using analog phones

On Monday, August 20, 2012, Ikka.vertika ikka.vert...@mitrakreasindo.com wrote: u need an analog telephone adaptor to connect your analog phone with switch hub. but the device is rather expensive. GrandStream 8 port ATA is about $200.00 which is much less than a single new SIP phone. Much much

Re: [asterisk-users] OT - Multi Function Printer with one touch scanning/emailing

On Fri, Jul 6, 2012 at 2:13 AM, Olivier oza_4...@yahoo.fr wrote: 2012/7/5, C F shma...@gmail.com: I searched a bit more, http://www.muratec.com/catalog/F320_config.html#email The above model supports t.37 That's very interesting to know. I quickly googled for t.37 and found several

Re: [asterisk-users] Amazon EC2 SIP floods - you can help

On Sun, Apr 18, 2010 at 12:25 PM, Randy R randulo2...@gmail.com wrote: On Sun, Apr 18, 2010 at 7:17 PM, Fred Posner f...@teamforrest.com wrote: There's also a link to it from the VoIP Tech Chat article. And we are also linking to Fred's original story which says it all about Amazon:

Re: [asterisk-users] wireless ATA

On Mon, May 4, 2009 at 2:36 PM, Jeff LaCoursierej...@jeff.net wrote: On Mon, 4 May 2009, Jimmy Godbout wrote: Take a look at wrp400 from Linksys/Cisco. It has 2 fxs, 802.11g and 4 switched ports + wan. Sigh, I suppose this is exactly what I was talking about :)  You guys sure know how to

Re: [asterisk-users] Video Conference

CrossPlatform Linux, Windows, Mac OpenSource WebHuddle at http://sourceforge.net/projects/webhuddle It has built in VOIP of some kind, don't remember the details. But why not use Asterisk or one of the free teleconference websites for the audio and WebHuddle for the webcams and desktop sharing.

Re: [asterisk-users] [asterisk-biz] T.38

On 10/17/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: On 10/17/07, Mike Hammett [EMAIL PROTECTED] wrote: I wasn't looking to involve Asterisk until I got it working solidly without it. ;-) I have yet to find on the entire internet a working example of T.38 pass-thru. I need to

Re: [asterisk-users] How to handle + prefix

i live in the US, but travel. i will second this ... Vodafone in Europe will give you a discount when you dial with a '+' instead of the local access code. Probably because Europeans roam in many foreign countries each with a different access code for long distance dialing. That would be messy,

Re: [asterisk-users] Ring the second line when 1st line is busy

Isn't this what you are looking for? http://voipspeak.net/index.php?option=com_contenttask=viewid=72Itemid=28 On 6/26/07, Deepak Naidu [EMAIL PROTECTED] wrote: Do any one any clue. This is what I need. I have a Polycom 501 phone, which support multiple lines ie on the LCD you can see the

Re: [asterisk-users] yum om centos

There is a yum command to install from cache and not look online. Details are in man yum.Never tried what you are doing, so i can't say for sure it would work. But you probably have to pass yum the makecache command on the first machine and install from cache on the second. But this is

Re: [asterisk-users] reset Polycom phones remotely

On 5/30/07, Mojo with Horan Company, LLC [EMAIL PROTECTED] wrote: An answer to your original question: if you can get someone _to_ the phones, on the polycom 30x's you would hold the VolDn, VolUp, DND, and Hold buttons for a while to reboot. For anyone with the 50x or 60x, you would hold the

Re: [asterisk-users] Vista compatibilty in SIP softphones

i try to get open source software that runs on Windows, OSX, and Linux. Java is often the best answer. Free and Open Source Java Client, but doesn't look like it supports Linux at the moment. http://sourceforge.net/projects/tudomais/ List of java based voip software https://voip.dev.java.net/

Re: [asterisk-users] VoiceXML + Nuance

Voxy - the only way to integrate VoiceXML applications in Asterisk. Configure your dial plan with the URL of your VoiceXML application and it's done. Is something the free and open source Voxy what you are looking for? http://sourceforge.net/projects/voxy On 5/3/07, wendell hamilton [EMAIL

[asterisk-users] Email to HP Product Suggestions - Seamless Transparent Fax Gateway

i sent a product suggestion to HP. It was a request to use software that already exists in their JetDirect and Multifunction Fax machines to make them seamlessly interoperate with a fax gateway in a way transparent to the end user. Essentially, giving the sysadmin a choice in fax transport

Re: [asterisk-users] SIP over VON

Like mentioned earlier, check the ports. Check latency and name resolution from both ends. There are many kinds of VPNs out there. Some VPNs are essentially one way. Depending on your system, you may have to establish a VPN session from the server back to the client. On 4/24/07, Ed Nuñez

Re: [asterisk-users] Purchasing a Sangoma A102 - should I get thehw echo cancellation or not?

A salesman told me that there are scenarios (analog vs T1 trunk lines) where echo cancellation will make things worse. Can anybody clear that up? On 4/23/07, Erik Anderson [EMAIL PROTECTED] wrote: On 4/23/07, Astawerks [EMAIL PROTECTED] wrote: You might as well spend an extra few bucks rather

Re: [asterisk-users] echo cancellation and ztdummy

Please tell me what hybrid echo is? Where does it come from? Does it have something to do with analog vs T1 trunk lines? On 4/23/07, William Moore [EMAIL PROTECTED] wrote: On 4/23/07, Patrick Fortin [EMAIL PROTECTED] wrote: Are echo cancellation parameters useful when using the ztdummy