August 2014 - App_rpt-users - lists.keekles.org

app_rpt requirements for Allstarlink.org WebTransceiver
by Bob 17 Aug '14

17 Aug '14

What are the app_rpt requirements for Allstarlink.org WebTransceiver to connect on opening? I have one working, and the other server/node, which I thought was a duplicate, is not. What am I missing? Thanks, Bob kk6ecm

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Re: [App_rpt-users] Web transceiver node display
by Bryan Fields 17 Aug '14

17 Aug '14

On 8/17/14, 12:20 PM, Tim Sawyer wrote: > Someone pointed out that SIP won’t work as a webtransceiver replacement > because SIP, unlike IAX, has no out of band signaling for PTT. > > I’m not sure how messaging works but I DO get a list of connected nodes on > my CISCO SIP phone when connected to a node on my Asterisk/app_rpt PBX. Hmm, good point :( Well I don't see an easy way to do it then. Anyone want to write a IAX2 client in javascript? http://tools.ietf.org/html/rfc5456 is the standard. Maybe the best way to do it is to modify what we have in Java. -- Bryan Fields 727-409-1194 - Voice 727-214-2508 - Fax http://bryanfields.net

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Re: [App_rpt-users] Web transceiver node display
by mike＠midnighteng.com 17 Aug '14

17 Aug '14

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Odd Telephone Portal / Bubble Map Issue
by Jon Byrne 17 Aug '14

17 Aug '14

Good Evening All My self and a few other node operators have noticed that when a user calls in via the telephone portal and connects to a node, the bubble map no longer works on that node, it shows it as if there are no connections. I have tested this on a number of hardware configurations and it does not seem to be node specific so may be an issue at the Bubble Map generating end, has anyone else noticed this issue or know who is best to talk to about it? It does not happen when a local sip connection is made to asterisk. Thanks for your help ------------ Jon Byrne 2E0RFU email(a)jonbyrne.com

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Re: [App_rpt-users] Questions about iax.conf
by mike＠midnighteng.com 17 Aug '14

17 Aug '14

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Re: [App_rpt-users] Web transceiver node display
by Bryan Fields 17 Aug '14

17 Aug '14

On 8/17/14, 10:59 AM, mike(a)midnighteng.com wrote: > > I would agree that a updated web-transceiver would be desirable, > That html5 ver is also java based. > > While java is simple and powerful, it also presents plenty of security problems. You're wrong :) It's Java Script and HTML5. Javascript has nothing to do with Java. It's confusing. I just tried it out with the online demo. Worked perfectly in firefox 31 on OSX for me with the SIP number http://sipml5.org/call.htm?svn=224# Private ID : 1 Pubic ID: sip:1@sip.allstarlink.org Passwd: allstar Realm: sip.allstarlink.org. Expert settings Disable Video X Enable RTCWeb Breaker X SIP outbound Proxy URL : udp://sip.allstarlink.org:5060 Disable Call Button Options X Reload the phone page and register. Put the node number (or 0) in the called number line and auth with your 10 digit pin. Looks like we just need some "speed dial buttons" and a text message interface. I don't know how that messaging is implemented. -- Bryan Fields 727-409-1194 - Voice 727-214-2508 - Fax http://bryanfields.net

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Questions about iax.conf
by ronald simpson 17 Aug '14

17 Aug '14

Hi guys, Is it possible to define what nodes on a multinode server use different codecs? For instance I want the following: Node 28412 to encode ilbc(I have recompiled to run this codec) Node 28413 to use ilbc/g726aal2 Node 28414 to use gsm Node 28415 to use gsm Node 28426 to use gsm I use these 5 nodes to monitor or talk out to many other larger node systems. When all are connected, the bandwidth goes up. I am trying to keep the bandwidth down as I have a very variable connect speed out here in the sticks. So the question is can each node be defined in iax.conf to codecs I want to use? Ron, N6GKJ

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Re: [App_rpt-users] Web transceiver node display
by mike＠midnighteng.com 17 Aug '14

17 Aug '14

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Re: [App_rpt-users] Web transceiver node display
by Bryan Fields 17 Aug '14

17 Aug '14

On 8/17/14, 12:57 AM, Tim Sawyer wrote: > Well, I thought I did. But all I have is the jar file. I’ll see if I can > rattle the Java tree and get it. At the risk of offending, perhaps it's time we look at moving away from the Java web client. My understanding is the Java client is basically a modified SIP client. There are SIP HTML5 clients out there we could build off of. Perhaps something like this: http://sipml5.org could be modified for our use? I'm not a programer, but I'd be willing to throw a donation to who ever can make it work to replace the current client. Thoughts? -- Bryan Fields 727-409-1194 - Voice 727-214-2508 - Fax http://bryanfields.net

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Issue with simplex node
by Wayne 16 Aug '14

16 Aug '14

HI all, Wondering if anyone may have experienced a similar problem. My simplex node operating duplex=1 (w/telemetry) will unkey and transmit in the middle of entering dtmf tones to it. For example as I'm enter *3node it will unkey at the star or sometimes at the 3 and sometimes at the end of the string at the last number and then sometimes it will work. Also, there are times I can be talking on the node and right in the middle of my transmission it will drop then come back when it goes back to receive. This has been baffling me for sometime. Setup: Desktop computer DMK URI Icom f6011 mobile Thanks Wayne Kh6mei --- This email is free from viruses and malware because avast! Antivirus protection is active. http://www.avast.com

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