October 2008 - App_rpt-users - lists.keekles.org

Re: [App_rpt] digium sources - makefile for app_rpt
by hwstar＠rodgers.sdcoxmail.com 09 Oct '08

09 Oct '08

Things have changed. You must enable app_rpt and radio_relax by using 'make menuselect' after running ./configure in the astrisk source tree. Also I personally don'r recommend running a radio system and a phone system on the same PC as your phone system will go down when you experiment with your radio system. If you are the only one in the houseold though, I suppose that's OK. Steve WA6ZFT > > From: MW0CQU <MW0CQU(a)ntlworld.com> > Date: 2008/10/09 Thu AM 06:51:34 PDT > To: app_rpt <app_rpt(a)lists.illiana.net> > Subject: [App_rpt] digium sources - makefile for app_rpt > > Hi all, > > I want to build my test app_rpt system on a "home" box wh ich is running > the excellent PBX_in_a_flash > asterisk distro. > > According to the guys in the PIAF forums the distro uses the stock > digium asterisk sources - which I can see as > I had to modify and re-compile a few lines of code to get the "R" button > working on my house phones. > > In preparation I have SCP'd all the relevant app_rpt and usbradio files > to my asterisk box, and went to un-comment > the line of code in the makefile to include app_rpt in the build at > runtime.... > > ...and there it wasn't??!! > > I have "cat *.* | grep app_rpt" in each sources folder and can find no > references whatsover to the aforementioned lines of code. > > my linux build is : > > 2.6.18-8.1.15.el5 #1 SMP Mon Oct 22 08:32:04 EDT 2007 i686 i686 i386 > GNU/Linux > > my asterisk build is: 1.4.15 > > Is anyone running app_rpt on a PIAF host please ? > Otherwise can someone clarify where exactly the code is in the makefile > (and which folder the makefile is in please) ? > > I have no room in the rack for another asterisk box! and I happen to > like PIAF rather a lot > > Many thanks > > 73 > > MW0CQU > > > > > > > _______________________________________________ > App_rpt mailing list > App_rpt(a)lists.illiana.net > http://lists.illiana.net/mailman/listinfo/app_rpt >

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[App_rpt] Asterisk Amateur Radio Conference Bridge
by Steven Henke 09 Oct '08

09 Oct '08

Hello Fellow Amateur Radio Over IP Enthusiasts, To help us communicate off-the-air, I've created a telephone conference bridge for us to use to discuss the project and use while troubleshooting installations. I will have this conference bridge activated for scheduled activities that will be posted via this list and whenever I am available to talk about amateur radio projects. There is no Public Switched Telephone Network (PSTN) access to this conference bridge. You must use VoIP to connect and here are a few ways you can do it. The first method is to use an Asterisk server by configuring routes in the dialplan. For example, the following entry in the /etc/asterisk/extensions_custom.conf file and its context [from-internal-custom] (or the equivalent file and context for your Asterisk system) routes calls dialed to extension 8501 to the XIPAR Conference Bridge via a SIP connection over the Internet. exten => 8501,1,Set(CALLERID(all)=YourCall<6785551234>) exten => 8501,n,Dial(SIP/8501(a)callin.xelatec.com) The second method also uses an Asterisk server and dialplan entries in /etc/asterisk/extensions_custom.conf and context [from-internal-custom] but makes the connection using the IAX2 protocol. exten => 8502,1,Set(CALLERID(all)=YourCall<6785551234>) exten => 8502,n,Dial(IAX2/guest(a)pbx.aretta.net/8502) As a third method you can also use the free Zoiper (http://www.zoiper.com) softphone client on a computer to connect directly to the Conference Bridge without going through an Asterisk server. Of course you must first have Zoiper installed and then add a new Zoiper IAX account with Account name 'callin', Server Hostname 'callin.xelatec.com', Username 'callin', and no password or other information. Select Show Advanced Options for that account and uncheck "Register on startup'. Apply the new account and click OK. Then from the main user interface, select the new account, go off hook and dial '8502'. That should connect you to the conference. And finally, you can use the iaxRpt (http://www.xelatec.com/xipar/iaxrpt) PTT Radio Dispatch Client to connect. From the iaxRpt menubar select Options, Accounts and Add. Then enter Account Name '8502', Description 'XIPAR Conference', Host 'callin.xelatec.com', Username 'callin' and you can leave 'Password and Confirm' blank. You can then select 8502 from the 'Dial Site/Node' drop down box and press 'Connect' to join the conference. Remember that you need to press the iaxRpt PTT switch when you want to talk! Whatever your method, when you successfully connect you will hear a greeting message that will tell you the status of the conference bridge. So that you can be ready for a conference, feel free to test your connection anytime you want. I hope to hear you soon! 73, Steven Henke, W9SH w9sh(a)xelatec.com

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[App_rpt] Dial-in access to remote base
by David McGough 09 Oct '08

09 Oct '08

Hi Everyone, I'm having a problem using my remote base via dial-in access to asterisk and I'm trying to figure out how to fix it.... On my asterisk box, I'm controlling 3 nodes currently. The first (node #21) is a UHF repeater, and the second (node #22) and third (node #23) nodes are simplex VHF and HF remote base radios....All these nodes are working fine, etc. The dial-in and autopatch functions work great with my repeater node (#21) ....But, for the remote base nodes, I can't key the rig from the SIP phone no matter what I try! I've tried all the phone modes: dumb, normal, simple and VOX mode from the extensions.conf. Nothing works. Here is a snippet of my extensions.conf: [radio-control] exten=21,1,Answer ; Works great! exten=21,n,Playback,rpt/node exten=21,n,Playback,digits/2 exten=21,n,Playback,digits/1 exten=21,n,Rpt,21|Pv exten=22,1,Answer ; Can't key the remote base exten=22,n,Playback,rpt/node exten=22,n,Playback,digits/2 exten=22,n,Playback,digits/2 exten=22,n,Rpt,22|Pv snippet of rpt.conf: [22] rxchannel = Radio/usb1 ; Rx audio/signalling channel duplex=0 remote=y remote_inact_timeout = 0 remote_timeout = 0 remote_timeout_warning = 0 remote_timeout_warning_freq = 0 voxtimeout=5000 ; time to let vox talk in ms voxrecover=500 ; time to take from vox one timed out in ms simplexpatchdelay=5 ; # of 20 ms frames to delay audio simplexphonedelay=5 ; # of 20 ms frames to delay audio [nodes] 21 = radio(a)127.0.0.1/21,NONE 22 = radio(a)127.0.0.1/22,NONE,y 23 = radio(a)127.0.0.1/23,NONE,y Thanks in advance! 73, David KB4FXC

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[App_rpt] digium sources - makefile for app_rpt
by MW0CQU 09 Oct '08

09 Oct '08

Hi all, I want to build my test app_rpt system on a "home" box which is running the excellent PBX_in_a_flash asterisk distro. According to the guys in the PIAF forums the distro uses the stock digium asterisk sources - which I can see as I had to modify and re-compile a few lines of code to get the "R" button working on my house phones. In preparation I have SCP'd all the relevant app_rpt and usbradio files to my asterisk box, and went to un-comment the line of code in the makefile to include app_rpt in the build at runtime.... ...and there it wasn't??!! I have "cat *.* | grep app_rpt" in each sources folder and can find no references whatsover to the aforementioned lines of code. my linux build is : 2.6.18-8.1.15.el5 #1 SMP Mon Oct 22 08:32:04 EDT 2007 i686 i686 i386 GNU/Linux my asterisk build is: 1.4.15 Is anyone running app_rpt on a PIAF host please ? Otherwise can someone clarify where exactly the code is in the makefile (and which folder the makefile is in please) ? I have no room in the rack for another asterisk box! and I happen to like PIAF rather a lot Many thanks 73 MW0CQU

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Re: [App_rpt] New ACID version uploaded to dl.allstarlink.org
by John Yaldwyn 07 Oct '08

07 Oct '08

Hi Steve, Apart from the new scripts and the echolink bits is there anything super important over 119 or should one just wait for the next time? Cheers, John ZL4JY -----Original Message----- From: app_rpt-bounces(a)lists.illiana.net [mailto:app_rpt-bounces@lists.illiana.net] On Behalf Of Stephen Rodgers Sent: Wednesday, 8 October 2008 08:36 To: Asterisk Repeater Controler Subject: Re: [App_rpt] New ACID version uploaded to dl.allstarlink.org James, A new ISO is not necessary, but all the scripts have drastically changed! astupd.sh depends on astinstall.sh in /etc/rc.d, and phase2.sh now calls astinstall.sh which is a common asterisk install script. There is a change to phase1.sh to remember where the download repository is located (acidrepo file), and everything keys off of this. So unless you want to figure out all the details of how the scripts work, your better off doing a fresh install! Just save all your conf files first so that you can re-install them over the default files after you run setup.sh and enable ssh access. Steve. James Nessen wrote: > So, should we download a new iso, or is there any way current nodes can get the new astupd.sh script? > > -----Original Message----- > From: app_rpt-bounces(a)lists.illiana.net > [mailto:app_rpt-bounces@lists.illiana.net] On Behalf Of Stephen > Rodgers > Sent: Sunday, October 05, 2008 1:31 PM > To: Asterisk Repeater Controler > Subject: [App_rpt] New ACID version uploaded to dl.allstarlink.org > > A new ACID version has been uploaded: > > 1. App_rpt version 0.151 supports chan_echolink and local messages 2. > Screen program added 3. chan_echolink channel driver 4. Asterisk > update script astupd.sh > > The current install iso has not been affected, but if you are in the > middle of an install, it might be a good idea to start over as the > scripts have changed. > > > Steve > WA6ZFT > _______________________________________________ > App_rpt mailing list > App_rpt(a)lists.illiana.net > http://lists.illiana.net/mailman/listinfo/app_rpt > _______________________________________________ > App_rpt mailing list > App_rpt(a)lists.illiana.net > http://lists.illiana.net/mailman/listinfo/app_rpt > _______________________________________________ App_rpt mailing list App_rpt(a)lists.illiana.net http://lists.illiana.net/mailman/listinfo/app_rpt

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[App_rpt] Error on asterisk box
by Jon Rorke 07 Oct '08

07 Oct '08

Hi All, I've been running a new node for week or so. The Box seems to work fine for a few hours , then an error occurs and the box is extremely slow to respond and can't recover. At this point it stops reporting too. It he CLI prompt this is the error reported: ERROR[5473]: chan_usbradio.c:1803 usbradio_read: HID process has died or something! The only way to get things going is to shut down the computer and reboot, then things are ok again for a few more hours. Any thoughts on what this might be? Is it a hardware failure or software? What I have done so far is to change the nic card - doesn't fix the problem. Moved the usb fob from the USB 2.0 PCI card to an internal usb port on the mother board - no change. Thinking of replacing hard drive and new fr4eash install with latest ACID CD. I'm not worried about this as its a new installation anyway. The box is a IBM P4 3Gig with 1 gig of mem.40 G drive. It is a referb unit so its not new. Thanks, Jon

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[App_rpt] app_rpt on other than x86pc hardware
by Eric Fort 06 Oct '08

06 Oct '08

Has app_rpt/(asterisk) ever been successfully compiled and run on non x86 hardware? The thought just came to mind that with the advent of the usb radio interface and the fact that asterisk is already running on some alternate firmwares for certain linksys (and other) consumer network devices that maybe one could inexpensively build an embeded node into such a device. Eric AF6EP

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[App_rpt] ACID install
by Eric Fort 06 Oct '08

06 Oct '08

Though ACID has been referenced before on this list, I don't remember it being defined (Allstar Centos Installation Disk). I'd always thought of the term in relation to databases. Is this then the preferred method of setting up a node? When giving notice that a new version is available please link to it: http://dl.allstarlink.org/. (ok my mailer truncated this in the original subject) Finally how does ACID compare to Limey Linux? Eric AF6EP On Sun, Oct 5, 2008 at 1:31 PM, Stephen Rodgers < hwstar(a)rodgers.sdcoxmail.com> wrote: > A new ACID version has been uploaded: > > 1. App_rpt version 0.151 supports chan_echolink and local messages > 2. Screen program added > 3. chan_echolink channel driver > 4. Asterisk update script astupd.sh > > The current install iso has not been affected, but if you are in the > middle of an install, it might be a good idea to start over as the > scripts have changed. > > > Steve > WA6ZFT > _______________________________________________ > App_rpt mailing list > App_rpt(a)lists.illiana.net > http://lists.illiana.net/mailman/listinfo/app_rpt >

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[App_rpt] Echolink_Channel
by Steve Zingman 06 Oct '08

06 Oct '08

Roger, Yes, it connects echolink to node 2151. To make a outgoing call use *33131002 to call me on echolink. *3 is connect to a node 3 says the following is a echolink node number 131002 is my node number. The echolink node number MUST be 6 digits. So a connect to echolink node 1234 would be: *33001234 Hope this helps. 73, Steve Roger F. Jordan Sr. wrote: > Thanks Steve, > > I have a few questions... > > Does the configuration you showed in your post below automagicly connect the > echolink node to your allstar node? > > If so... > > What is the procedure to DTMF to an Echolink node? > > > Roger, > There is no change to the configuration in rpt.conf > Everything is in /etc/asterisk/echolink.conf and extensions.conf > > echolink.conf: > [el0] > confmode=no > ipaddr=0.0.0.0 > port=5198 > call=K4NRG-R > pwd=thisismyrealpassword > name=Nerd Repeater Group > qth=Allstar node 2151 > email=k4nrg(a)qsl.net > maxstns=5 > rtcptimeout=10 > node=131002 > recfile=/tmp/recorded.gsm > server1=server1.echolink.org > server2=server2.echolink.org > server3=server3.echolink.org > astnode=2151 > context=echolink-in > > > extensions.conf: > [echolink-in] > exten => 2151,1,rpt(2151|X) > > 73, Steve N4IRS >

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Re: [App_rpt] App_rpt Digest, Vol 32, Issue 4
by Steve Zingman 05 Oct '08

05 Oct '08

> ---------------------------------------------------------------------- > > Message: 1 > Date: Sat, 4 Oct 2008 17:27:21 -0400 > From: "Roger F. Jordan Sr." <w4rfj(a)comcast.net> > Subject: [App_rpt] Echolink_Channel > To: "'Asterisk Repeater Controler'" <app_rpt(a)lists.illiana.net> > Message-ID: <mailman.29.1223183942.2534.app_rpt(a)lists.illiana.net> > Content-Type: text/plain; charset="us-ascii" > > Hi. > > > > I am looking for info or examples as to how to set up the Echolink_Channel > in the rpt.conf file. > > > > Thanks, > > Roger W4RFJ > > Roger, There is no change to the configuration in rpt.conf Everything is in /etc/asterisk/echolink.conf and extensions.conf echolink.conf: [el0] confmode=no ipaddr=0.0.0.0 port=5198 call=K4NRG-R pwd=thisismyrealpassword name=Nerd Repeater Group qth=Allstar node 2151 email=k4nrg(a)qsl.net maxstns=5 rtcptimeout=10 node=131002 recfile=/tmp/recorded.gsm server1=server1.echolink.org server2=server2.echolink.org server3=server3.echolink.org astnode=2151 context=echolink-in extensions.conf: [echolink-in] exten => 2151,1,rpt(2151|X) 73, Steve N4IRS

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