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Re: [App_rpt] Asterisk Amateur Radio Conference Bridge
by John Yaldwyn 10 Oct '08

10 Oct '08

Hi Steve, Got the message just fine via IAX2 from New Zealand! Only received silence to the SIP connection, debug stated: -- SIP/callin.xelatec.com-.... answered SIP/821-... -- Native bridging SIP/821-.... and SIP/callin.xelatec.com-... Cheers, John ________________________________ From: app_rpt-bounces(a)lists.illiana.net on behalf of Steven Henke Sent: Fri 10/10/2008 5:15 AM To: app_rpt(a)lists.illiana.net Subject: [App_rpt] Asterisk Amateur Radio Conference Bridge 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 _______________________________________________ App_rpt mailing list App_rpt(a)lists.illiana.net http://lists.illiana.net/mailman/listinfo/app_rpt

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Re: [App_rpt] Asterisk Amateur Radio Conference Bridge
by Tony Langdon, VK3JED 10 Oct '08

10 Oct '08

At 03:15 AM 10/10/2008, you wrote: >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. Works a treat, got the greeting. I've programmed it into my VoIP provider's speed dials (they give me a lot of the functionality of an Asterisk box), so I can use it from my regular IP phone. :) I can't change my caller ID string for that one connection. Whether we'll be able to catch up due to the time difference is another thing. :) 73 de VK3JED http://vkradio.com

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Re: [App_rpt] need acid boot disk (floppy img)
by John Yaldwyn 10 Oct '08

10 Oct '08

Hi Eric, I posted about CPU upgrade for your system there's a good page here that migh be of interest: http://homepage.hispeed.ch/rscheidegger/p2b_procupgrade_faq.html Re the boot problem, lots of people have this with older machines. The first step is to make certain the CDROM drive is attached to a motherboard IDE header and not an add in card. Try putting the CDROM as the slave on primary IDE channel. Then set the BIOS to boot first floppy, then CDROM, then HDD. If all that looks good, can you boot any other bootable CDs like and old Win95, Win2K, etc, install CD that you migh have handy? John ________________________________ From: app_rpt-bounces(a)lists.illiana.net [mailto:app_rpt-bounces@lists.illiana.net] On Behalf Of Eric Fort Sent: Friday, 10 October 2008 15:10 To: Asterisk Repeater Controler Subject: [App_rpt] need acid boot disk (floppy img) I'd like to install ACID on a asus P2B-D dual P3 system I have. unfortunately it will not boot from cd so I need to use a boot floppy. What would you suggest for a boot image that I can rawwite to a 1.44M floppy and bootstrap the cd? Thanks, Eric

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[App_rpt] need acid boot disk (floppy img)
by Eric Fort 10 Oct '08

10 Oct '08

I'd like to install ACID on a asus P2B-D dual P3 system I have. unfortunately it will not boot from cd so I need to use a boot floppy. What would you suggest for a boot image that I can rawwite to a 1.44M floppy and bootstrap the cd? Thanks, Eric

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Re: [App_rpt] Dial-in access to remote base
by hwstar＠rodgers.sdcoxmail.com 10 Oct '08

10 Oct '08

David, app_rpt.c version 0.152 appears to support setting idtime=0 and id messages won' be queued: 10985 /* place an ID event in the telemetry queue */ 10986 10987 static void queue_id(struct rpt *myrpt) 10988 { 10989 if(myrpt->p.idtime){ /* ID time must be non-zero */ 10990 myrpt->mustid = myrpt->tailid = 0; 10991 myrpt->idtimer = myrpt->p.idtime; /* Reset our ID timer */ 10992 rpt_mutex_unlock(&myrpt->lock); 10993 rpt_telemetry(myrpt,ID,NULL); 10994 rpt_mutex_lock(&myrpt->lock); 10995 } 10996 } 10997 The source sometimes ends up being the user manual. Steve WA6ZFT > > From: David McGough <kb4fxc(a)inttek.net> > Date: 2008/10/09 Thu PM 04:05:46 PDT > To: Asterisk Repeater Controler <app_rpt(a)lists.illiana.net> > Subject: Re: [App_rpt] Dial-in access to remote base > > > Nope, the idea of setting duplex=1 doesn't work. With duplex=1, the radio > TX is ALWAYS briefly keyed immediately after the COS signal from RX drops. > I tried setting hangtime=0, etc. This causes my HF rig to go into an > infinite key/unkey loop due to the nature of its voice squelch circuit. > > I guess I can just make an audio file of silence for the ID and leave > duplex=0..... > > 73, David KB4FXC > > > On Thu, 9 Oct 2008, David McGough wrote: > > > > > > > ...I just found something else of interest. When I copied my repeater > > stanza to my original remote base nodes, I left the original CW ID audio > > file specified as the object of the idrecording variable. On the asterisk > > console, I'm seeing the standard "Playing 'kb4fxc.4.ulaw'" message that I > > get for the repeater node on all the nodes now. And, if the remote base > > (now a standard node) is keyed, the ID is transmitted! If unkeyed, it's > > not transmitted. Note that duplex=0 for these nodes. I guess I can make a > > audio file of "silence" for these nodes, etc. Or, perhaps, make a correct > > ID file and set duplex=1??? > > > > 73, David KB4FXC > > > > > > On Thu, 9 Oct 2008, Eric Fort wrote: > > > > > on a side note could the frequency agility be done through the dialplan and > > > agi? > > > > > > Eric > > > > > > On Thu, Oct 9, 2008 at 12:57 PM, David McGough <kb4fxc(a)inttek.net> wrote: > > > > > > > > > > > Hi Steve, > > > > > > > > Okay, I reconfigured both remote bases as standard nodes--I just copied > > > > the context used for my repeater node and set duplex=0. This works > > > > properly with VOX mode, manual keying/unkeying, etc., just like with the > > > > repeater node....No issues. > > > > > > > > And, okay about the link status--I just thought the segfault was > > > > interesting. > > > > > > > > So, other than loss of frequency agility, can you think of any other > > > > "show stoppers" that I might hit by retiring remote base mode entirely and > > > > just using standard mode for everything?? > > > > > > > > 73, David KB4FXC > > > > > > > > > > > > > > > > On Thu, 9 Oct 2008 hwstar(a)rodgers.sdcoxmail.com wrote: > > > > > > > > > See below > > > > > > > > > > Steve > > > > > WA6ZFT > > > > > > > > > > > > > > > > > > > <* snip *> > > > > > > > > > > > > > > First off: Link status doesn't make sense on a remote base! > > > > > Remote bases are for exclusive use by one user at a time! > > > > > > > > > > Key-unkey flakiness. Two things to check: > > > > > > > > > > 1. What happens when you configure the HF and VHF radios as a standard > > > > node (with duplex=0)? The remote base code is separate from the standard > > > > node code, and there could be a keying bug could be in the remote base code. > > > > > > > > > > 2. Please confirm the phone key/unkey is not flakey on your standard > > > > node.This key/unkey issue could be a SIP configuration issue. > > > > > > > > > > > > > > > > > > _______________________________________________ > > > > 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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Re: [App_rpt] Dial-in access to remote base
by hwstar＠rodgers.sdcoxmail.com 09 Oct '08

09 Oct '08

See below Steve WA6ZFT > > From: David McGough <kb4fxc(a)inttek.net> > Date: 2008/10/09 Thu AM 10:56:12 PDT > To: Asterisk Repeater Controler <app_rpt(a)lists.illiana.net> > Subject: Re: [App_rpt] Dial-in access to remote base > > > > Hi Steve, > > Thanks for the help! I'm just about there now, I think! I added the config > statements you suggested and now I can key the remote base via '*9' and > unkey via '#'.....But this seems a bit flakey--I can get the remote base > keyed, but, it doesn't always unkey..Also, VOX mode still doesn't want to > work. > > Some other observations are: I noticed that if I use simple mode, once the > remote base is keyed it won't unkey....Not really an issue to me, but, an > observation. And, function *7 (link status) causes a seg fault--should > that happen? First off: Link status doesn't make sense on a remote base! Remote bases are for exclusive use by one user at a time! Key-unkey flakiness. Two things to check: 1. What happens when you configure the HF and VHF radios as a standard node (with duplex=0)? The remote base code is separate from the standard node code, and there could be a keying bug could be in the remote base code. 2. Please confirm the phone key/unkey is not flakey on your standard node.This key/unkey issue could be a SIP configuration issue. > > Please note as I mentioned earlier that VOX, etc., works fine on the > repeater node (#21), interestingly.... > > Here are my current configs: > > rpt.conf > -------- > > [22] ; Change this to your assigned node number > rxchannel = Radio/usb1 ; Rx audio/signalling channel > duplex=0 > remote=y > functions = functions_remote > phone_functions = functions_remote > authlevel = 0 ; should be set to 0 or left undefined. > 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=10 ; # of 20 ms frames to delay audio > simplexphonedelay=10 ; # of 20 ms frames to delay audio > > [functions_remote] > 7=ilink,5 ; Link status > 8=status,2 ; Time > 9=cop,6 ; Phone TX key > > > extensions.conf > ---------------- > > [radio-control] > exten=22,1,Answer > exten=22,n,Playback,rpt/node > exten=22,n,Playback,digits/2 > exten=22,n,Playback,digits/2 > exten=22,n,Rpt,22|Pv > > > > > On Thu, 9 Oct 2008 hwstar(a)rodgers.sdcoxmail.com wrote: > > > > > > > Do you have a functions_remote stanza and is phone_functions pointing > to a remote_functions function table? Do you have a 98=cop,6 statement in > the remote_functions stanza? > > > > [functions_remote] > > . > > . > > . > > 98=cop,6 > > > > Finally, is authlevel set to 0 or undefined in the node stanza? If > authlevel is >0, you won't be able to key anything and 20 seconds later, > you'll be disconnected! > > > > > > [XXXX] > > > > rxchannel = Zap/2 > > remote = ft897 > > functions = remote-functions > > phone_functions = remote-functions ; important! > > authlevel = 0; should be set to 0 or left undefined. > > > > > > Steve > > WA6ZFT > > > > > > > > > > From: David McGough <kb4fxc(a)inttek.net> > > > Date: 2008/10/09 Thu AM 07:53:24 PDT > > > To: app_rpt(a)lists.illiana.net > > > Subject: [App_rpt] Dial-in access to remote base > > > > > > > > > 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 > > > > > > > > > > > > > > > _______________________________________________ > > > 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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Re: [App_rpt] Dial-in access to remote base
by hwstar＠rodgers.sdcoxmail.com 09 Oct '08

09 Oct '08

Do you have a functions_remote stanza and is phone_functions pointing to a remote_functions function table? Do you have a 98=cop,6 statement in the remote_functions stanza? [functions_remote] . . . 98=cop,6 Finally, is authlevel set to 0 or undefined in the node stanza? If authlevel is >0, you won't be able to key anything and 20 seconds later, you'll be disconnected! [XXXX] rxchannel = Zap/2 remote = ft897 functions = remote-functions phone_functions = remote-functions ; important! authlevel = 0; should be set to 0 or left undefined. Steve WA6ZFT > > From: David McGough <kb4fxc(a)inttek.net> > Date: 2008/10/09 Thu AM 07:53:24 PDT > To: app_rpt(a)lists.illiana.net > Subject: [App_rpt] Dial-in access to remote base > > > 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 > > > > > _______________________________________________ > 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 Correction
by Steven Henke 09 Oct '08

09 Oct '08

Oops. My cut and paste mistake from another conference bridge! The IAX2 connection syntax should read: exten => 8502,1,Set(CALLERID(all)=YourCall<6785551234>) exten => 8502,n,Dial(IAX2/callin(a)callin.xelatec.com/8502) If you need a hosted Asterisk solution, Aretta Comms is a great choice! Steve, W9SH

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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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