- App_rpt-users - lists.keekles.org

Re: [App_rpt-users] 64 bit
by hwstar＠rodgers.sdcoxmail.com 10 Feb '10

10 Feb '10

> > From: "Ken" <ke2n(a)cs.com> > Date: 2010/02/09 Tue AM 10:21:49 PST > To: "'Steve Gladden'" <steve(a)michiganbroadband.com> > CC: app_rpt-users(a)qrvc.com > Subject: Re: [App_rpt-users] 64 bit > > Thanks Steve - see my question below > > > -----Original Message----- > > From: Steve Gladden [mailto:steve@michiganbroadband.com] > > Sent: Tuesday, February 09, 2010 11:38 AM > > To: Ken > > Cc: app_rpt-users(a)qrvc.com > > Subject: Re: [App_rpt-users] 64 bit > > > > Hello Ken, > > I've done it in the past by copying all of the scripts from > > a live 32 bit system over to the 64 bit system. > > I first did all of the Centos updates on the 64 bit system.. > > > > Then I ran the stage2 script on the 64bit system.. > > ===== > I would probably need more help with this than you could provide- > But, can you clarify - does this result in a "normal" 32 bit APP_RPT running > under the 64 bit OS? > > Or does it use the 64 bit compiler, creating a 64-bit version of the > "forked" Asterisk code? > (This latter one sounds like it has some risk of not working). > > Ken Jamrogowicz > > _______________________________________________ > App_rpt-users mailing list > App_rpt-users(a)qrvc.com > http://qrvc.com/mailman/listinfo/app_rpt-users > You'll get 64 bit code of the forked version it it compiles without errors. It may work or it may not work I don't know. You would be on your own and we would not be able to support such a build as it is outside of the scope of the project. Steve WA6ZFT

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64 bit
by Ken 09 Feb '10

09 Feb '10

I know that CentOS and Asterisk are available in 64-bit versions I have a 64 bit machine. Is it possible to make an install of app_rpt that runs in 64 bit? if so, how? Using the standard ISO and install procedure, it installs 32 bit OS and modules. Thanks Ken Jamrogowicz

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individual graphical node status
by Randy Hammock 08 Feb '10

08 Feb '10

Is there a way to get an individual Graphical Node Status Page? For example, for a local web page, to show a "Zoloft" diagram ( as Jim calls it) for the nodes connected to 2035 without the rest of the world. -- Randy Hammock KC6HUR http://kc6hur.net IRLP 4494 / ALLSTAR 2035 http://irlp.kc6hur.net

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Calling into the node operator's local SIP phone
by Keith Williamson 08 Feb '10

08 Feb '10

Hi, Several people have expressed an interest in how to configure Allstar to allow radio users to connect to the node-operator's local SIP phone. It turns out it's pretty easy (once you have a local SIP phone configured of course). To do this without configuring more general autopatch access with it's potential risks, you can create a specific context for just allowing a radio user to access one "outbound" SIP connection, the local SIP phone. In rpt.conf, I uncommented the "autopatchdn" function and duplicated and uncommented the "autopatchup" function. These two functions are in the [functions] stanza available to radio users. In the new "autopatchup" function, I changed the default DTMF string from 6 to 61 and added the "context=" option. I set the option to "context=node-op". So the function now looks like this: 61=autopatchup,context=node-op,noct=1,farenddisconnect=1,dialtime=20000 Then in extensions.conf, I added a stanza for [node-op]: [node-op] exten => 1,1,Answer exten => 1,n,Dial(SIP/200,10) exten => 1,n,Playback(vm-nobodyavail) exten => 1,n,Hangup Change the SIP/200 above to SIP/whatever-your-local-extension-is. Since we modified rpt.conf, you need to restart asterisk. Now, if the radio user keys in *611, the autopatch will be invoked and extension "1" in context [node-op] will be called where it will be answered and then will dial the local SIP phone extension. If you don't pickup, it will timeout, play the "nobody available to take your call" message, and hangup. Of course you can integrate this into a full autopatch configuration by modifying the default dialplan in context "radio" but I'm not willing to open my node up to dialing out to the world (yet). 73's, Keith KF7DRV

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Re: [App_rpt-users] Calling into the node operator's local SIP phone
by Keith Williamson 07 Feb '10

07 Feb '10

On Sun, Feb 7, 2010 at 1:53 PM, Stephen - K1LNX <k1lnx(a)k1lnx.net> wrote: > Keith, > This works like a champ! I just so happened to re-flash one of my Cisco > phones back to SIP this afternoon for playing with and the timing could not > have been more perfect lol. A very very useful feature! > > 73 > Stephen > K1LNX > > > On Sun, Feb 7, 2010 at 2:29 PM, Keith Williamson <hkwilliamson(a)gmail.com>wrote: > >> Hi, >> >> Several people have expressed an interest in how to configure Allstar to >> allow radio users to connect to the node-operator's local SIP phone. It >> turns out it's pretty easy (once you have a local SIP phone configured of >> course). To do this without configuring more general autopatch access with >> it's potential risks, you can create a specific context for just allowing a >> radio user to access one "outbound" SIP connection, the local SIP phone. In >> rpt.conf, I uncommented the "autopatchdn" function and duplicated and >> uncommented the "autopatchup" function. These two functions are in the >> [functions] stanza available to radio users. In the new "autopatchup" >> function, I changed the default DTMF string from 6 to 61 and added the >> "context=" option. I set the option to "context=node-op". So the function >> now looks like this: >> >> 61=autopatchup,context=node-op,noct=1,farenddisconnect=1,dialtime=20000 >> >> Then in extensions.conf, I added a stanza for [node-op]: >> >> [node-op] >> exten => 1,1,Answer >> exten => 1,n,Dial(SIP/200,10) >> exten => 1,n,Playback(vm-nobodyavail) >> exten => 1,n,Hangup >> >> Change the SIP/200 above to SIP/whatever-your-local-extension-is. Since we >> modified rpt.conf, you need to restart asterisk. >> >> Now, if the radio user keys in *611, the autopatch will be invoked and >> extension "1" in context [node-op] will be called where it will be answered >> and then will dial the local SIP phone extension. If you don't pickup, it >> will timeout, play the "nobody available to take your call" message, and >> hangup. >> >> Of course you can integrate this into a full autopatch configuration by >> modifying the default dialplan in context "radio" but I'm not willing to >> open my node up to dialing out to the world (yet). >> >> 73's, >> >> Keith >> KF7DRV >> >> >> _______________________________________________ >> App_rpt-users mailing list >> App_rpt-users(a)qrvc.com >> http://qrvc.com/mailman/listinfo/app_rpt-users >> >> > > > -- > ********************************** > Stephen Brown - ARS K1LNX > Johnson City, TN EM86 > http://www.k1lnx.net > google voice: 423-665-9367 > ********************************** > Yeah, isn't allstar/asterisk the greatest? It's like Lego's for computers, radios, and telephones. Cheers, Keith

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Problems with Cepstral and Festival
by Chuck Kraly 07 Feb '10

07 Feb '10

Hi Guys, I have been trying to get Cepstral and Festival to both or either play with ACID. The installs go well, and I can execue them from CLI. The pc hesitates while it executes, but I get no sound what so ever out of either the USB fob, or local sound connection. If somoen on the list has time, could you email me at ckraly(a)gmail.com and help me see what I might be missing? Thanks Chuck K0XM

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Private Network
by Monty Dana 04 Feb '10

04 Feb '10

Hello Everyone! I wondering how one might set up a private network of connected nodes? Right now I have two nodes setup here at my home, and would like to experiment with having them both tied together on my private home network, and still have one node able to access allstar and echolink, is that possible? What the main reason for this question is that many of you know, broadband is hard to come by on mountain tops and if it is available very expensive. So we are looking for other alternatives, ie Wifi links, spread spectrum T1, etc to have a private network among sites with a single public IP address, like a home network. Thanks, Monty KE7JVX

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Example of how to setup a sip phone for reverse autopatch
by Keith Williamson 02 Feb '10

02 Feb '10

Thought I'd share this thread with the list since it might be helpful to someone trying to setup a local sip phone for reverse autopatch. Bob and I went off the list when we first got started on this because I was afraid the thread could get quite lengthy if we ran into the (almost) inevitable problems. However, everything worked the first time through a combination of luck on my part explaining things (blue moon) and Bob's excellent ability to read, interpret, and follow my drivel. Bob, if there were things I explained incorrectly below that you figured your way around, please chime in so that the next unwary ham who tries this will be forewarned. Also note that some of this is specific to Bob's SIP phone but in looking through the manual for the phone, I found it to be pretty typical of other SIP phones I've worked with. Finally, apologies for the reverse ordering of the thread versus the way we normally (try) to order thread (newest reply at bottom). 73's Keith KF7DRV Allstar Node 2541 ---------- Forwarded message ---------- From: Bob Brown - WØNQX <bbrown(a)byrg.net> Date: Sun, Jan 31, 2010 at 12:03 PM Subject: SIP IP pHONE config - was: Connect notifications via AGI or System scripts To: Keith Williamson <hkwilliamson(a)gmail.com> hi keith got the first part done ~~~~~~~~~~~~~~~~~~~~~~ sip show peers Name/username Host Dyn Nat ACL Port Status 9009/9009 10.0.0.40 D 5060 Unmonitored 1 sip peers [Monitored: 0 online, 0 offline Unmonitored: 1 online, 0 offline] ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ whats next? -- Thanks in Advance Bob Brown, WØNQX Kansas City Metro Area http://sm0kenet.net http://byrg.net http://kcdstar.byrg.net Quis custodiet ipsos custodes? On Sun, Jan 31, 2010 at 12:09, Keith Williamson <hkwilliamson(a)gmail.com>wrote: > OK great. Next step is to get the phone registered with the Allstar node. > I'm going to assume the phone is connected to the internal network the node > is on and for the purposes of this explanation I'll say the internal IP of > your Allstar node is 192.168.0.1 (and that that IP is either static or gets > a static lease via DHCP). Obviously, substitute your node's actual internal > IP. > > In the web interface for the phone, enter the Allstar node internal IP in > ALL of the following fields of Line 1 settings for Profile 1: > > SIP Proxy Server > Outbound Proxy Server > Registrar Server > Outbound Registrar Server > > Leave the SIP Proxy Server Port and Registrar Proxy Server Port set at the > default of 5060. > > Also in the same Line 1 Profile 1 menu, you need to set the phone's > extension info. You can make this whatever you want but for the purposes of > this explanation we'll call it Bobs208. Set Bobs208 in the following fields: > > Phone Number > User Name > Authorized ID > > Finally, you need to set a password for authentication. We'll use > "secret123". Set the following field to secret123: > > Authorized Password > > This should be enough on the phone side of things so now down at the bottom > of the menu, do Save Settings followed by Logout. > > Now we move over to the Allstar node. Open /etc/asterisk/sip.conf with your > favorite editor (I'm a vi guy myself) and add a stanza for your phone (below > the end of the [general] stanza) that would be something like this: > > [Bobs208] > type=friend > secret=secret123 > context=radio-control > host=dynamic > disallow=all > allow=ulaw > transfer=no > > Now save the changes to sip.conf and exit the editor. Next go into the CLI > (e.g. asterisk -vr) and type "sip reload" to force asterisk to reread the > sip.conf file. At this point, asterisk should be ready register your phone. > > Probably the most dependable way to get the phone to activate the changes > you made to the phone's SIP parameters and to force the phone to attempt to > register with Allstar/Asterisk is to just power-cycle the phone. > > I haven't read enough about this phone to know how the phone indicates it's > registration state after it finishes booting but you can watch the Allstar > CLI and you should be able to see the registration occuring. You can also > type the following CLI command to check if you phone is registered: > > CLI> sip show peers > > Let me know how all this goes and then we can proceed to the final step > once the phone is registered. > > 73's > > Keith > > KF7DRV > > > > > > On Sat, Jan 30, 2010 at 11:57 PM, Bob Brown - WØNQX <bbrown(a)byrg.net>wrote: > >> ya mine is a virgin fone and the default is 1234 >> >> i got this from a ip phone vendor/wholsaler it was a freebie he says it >> was sip compatable >> >> i have managed to log into it from the web interface and from the admin >> menu in the front panle. >> >> >> >> -- >> Thanks in Advance >> >> Bob Brown, WØNQX >> >> Kansas City Metro Area >> >> http://sm0kenet.net >> >> http://byrg.net >> >> http://kcdstar.byrg.net >> >> Quis custodiet ipsos custodes? >> >> >> >> On Sat, Jan 30, 2010 at 22:45, Keith Williamson <hkwilliamson(a)gmail.com >> > wrote: >> >>> Hi Bob, >>> >>> I found the manual for your T207/T208 phone..hopefully you have that too. >>> Getting the phone registered with Asterisk requires that the phone be an "S" >>> model (T207S or T208S) which means it has SIP firmware. The T207M and T208M >>> models run MGCP which isn't compatible with Asterisk. The other requirement >>> is that you need to know the admin password. If it's never been changed it >>> should be the default "1234". You can configure the phone settings either >>> directly on the phone or via your web browser (although you need to at least >>> configure the TCP/IP parameters using the phone menus before you can switch >>> to using the more convenient web interface). >>> >>> Perhaps you have already done this part and have the phone connected via >>> Ethernet to the network the node is on? >>> >>> Let me know and we can go from there. >>> >>> 73s, >>> >>> Keith >>> >>> >>> On Sat, Jan 30, 2010 at 10:02 AM, Bob Brown - WØNQX <bbrown(a)byrg.net>wrote: >>> >>>> my ipphone: TSIPP2008BG-R1 >>>> >>>> sure any help would be great! >>>> >>>> >>>> >>>> -- >>>> Thanks in Advance >>>> >>>> Bob Brown, WØNQX >>>> >>>> Kansas City Metro Area >>>> >>>> http://sm0kenet.net >>>> >>>> http://byrg.net >>>> >>>> http://kcdstar.byrg.net >>>> >>>> Quis custodiet ipsos custodes? >>>> >>>> >>>> >>>> On Fri, Jan 29, 2010 at 23:01, Keith Williamson < >>>> hkwilliamson(a)gmail.com> wrote: >>>> >>>>> Hi Bob, >>>>> >>>>> No problem. What make/model of ipphone is it? The first thing required >>>>> is a stanza in sip.conf defining the extension, user, password, etc >>>>> for that particular phone. The required elements vary from phone to >>>>> phone so let me know what kind you have. After creating the entry in >>>>> sip.conf, you just do a "sip reload" at the CLI and check to see if >>>>> the phone gets registered OK. Once you have that, you just need to add >>>>> a stanza to extensions.conf to enable the reverse autopatch. >>>>> >>>>> There's a pretty complete description of what's required in the ACID >>>>> Admin Manual. >>>>> >>>>> Cheers, >>>>> >>>>> Keith >>>>> >>>>> >>>>> On 1/29/10, Bob Brown - WØNQX <bbrown(a)byrg.net> wrote: >>>>> > hi keith >>>>> > >>>>> > i would be interrested in your config files to set up the ipphone >>>>> > on your node >>>>> > >>>>> > i have a 4 line ip desk phone i would like to set up to do what you >>>>> describe >>>>> > >>>>> > i am not very fluent in asterisk set up for ip fones. >>>>> > >>>>> > would you be willing to share? >>>>> > >>>>> > >>>>> > -- >>>>> > Thanks in Advance >>>>> > >>>>> > Bob Brown, WØNQX >>>>> > >>>>> > Kansas City Metro Area >>>>> > >>>>> > http://sm0kenet.net >>>>> > >>>>> > http://byrg.net >>>>> > >>>>> > http://kcdstar.byrg.net >>>>> > >>>>> > Quis custodiet ipsos custodes? >>>>> > >>>>> > >>>>> > >>>>> > On Fri, Jan 29, 2010 at 22:04, Keith Williamson >>>>> > <hkwilliamson(a)gmail.com>wrote: >>>>> > >>>>> >> Hi, >>>>> >> >>>>> >> Recently, I got reverse autopatch configured on my simplex node >>>>> (2541) and >>>>> >> everything works great. In the shack, I have a Polycom IP501 speaker >>>>> phone >>>>> >> that I use to monitor or connect into QSO's on the node. Out of the >>>>> shack, >>>>> >> I >>>>> >> can call into the node using my Blackberry and do the same. So the >>>>> next >>>>> >> challenge is to get quick notifications of users connecting into the >>>>> node >>>>> >> either via radio or Internet (Echolink and Allstar..no IRLP yet). I >>>>> >> created >>>>> >> a pair of Twitter accounts; one for the node and another for me >>>>> >> personally. >>>>> >> I added my node's Twitter account to the ones I follow with my >>>>> personal >>>>> >> account and created an AGI scripts that formats a curl command which >>>>> posts >>>>> >> some of the context variables passed in when Asterisk invokes the >>>>> script >>>>> >> (context name, extension, callerid, etc). I then added the AGI call >>>>> into >>>>> >> various extension stanzas in extensions.conf to test. If, for >>>>> instance, I >>>>> >> do >>>>> >> a reverse autopatch and connect to the reverse autopatch >>>>> "extension", I >>>>> >> almost immediately get an SMS tweet on the Blackberry. Great. >>>>> However, it >>>>> >> seems that connections via the radio are not processed in any way in >>>>> >> extensions.conf (thought [default] stanza would apply but apparently >>>>> only >>>>> >> applies to autopatch), and inbound connections from either Allstar >>>>> or >>>>> >> Echolink, while processed through [radio-secure], hang on the AGI >>>>> call and >>>>> >> don't proceed to the following call to rpt. I'm assuming this is >>>>> because >>>>> >> AGI >>>>> >> calls generally follow an "Answer" and are only valid in the >>>>> connected >>>>> >> state. I tried "deadAGI" since it doesn't seems to be dependent on >>>>> being >>>>> >> in >>>>> >> the answered state but it hung the same way. It's possible I've got >>>>> a >>>>> >> problem in the AGI script which is causing the hang but it certainly >>>>> >> doesn't >>>>> >> occur when the AGI call follows "Answer". >>>>> >> >>>>> >> So I'm looking for help to understand other possible ways to invoke >>>>> a >>>>> >> script that calls curl to post the tweet when a radio user makes an >>>>> >> outbound >>>>> >> connection to Allstar or Echolink and when an Echolink or Allstar >>>>> user >>>>> >> connects in to the radio. >>>>> >> >>>>> >> Ideas? >>>>> >> >>>>> >> Thanks and 73's >>>>> >> >>>>> >> Keith >>>>> >> >>>>> >> KF7DRV >>>>> >> >>>>> >> _______________________________________________ >>>>> >> App_rpt-users mailing list >>>>> >> App_rpt-users(a)qrvc.com >>>>> >> http://qrvc.com/mailman/listinfo/app_rpt-users >>>>> >> >>>>> >> >>>>> > >>>>> >>>>> -- >>>>> Sent from my mobile device >>>>> >>>> >>>> >>> >> >

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Reverse Autopatch radio PTT
by Stephen - K1LNX 02 Feb '10

02 Feb '10

I finally got my main PBX peered to my app_rpt box (long story) and I set up normal phone control mode to have full functionality. With simple phone control, I can key/unkey the transmitter with * and # respectively. How do I accomplish that in normal phone control mode? Also, if I connect to an echolink or allstar node from my phone, how does audio (if it can be) get sent to the network? For example, say I connect to my node via phone and I want to connect to Allstar node 2000. Can I send audio from my phone? I am finding it very useful to monitor things, but the ability to transmit audio to the network would be rather neat..... Is this possible or is this beyond the scope of design? tnx and 73 Stephen K1LNX -- ********************************** Stephen Brown - ARS K1LNX Johnson City, TN EM86 http://www.k1lnx.net google voice: 423-665-9367 **********************************

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Connect notifications via AGI or System scripts
by Keith Williamson 30 Jan '10

30 Jan '10

Hi, Recently, I got reverse autopatch configured on my simplex node (2541) and everything works great. In the shack, I have a Polycom IP501 speaker phone that I use to monitor or connect into QSO's on the node. Out of the shack, I can call into the node using my Blackberry and do the same. So the next challenge is to get quick notifications of users connecting into the node either via radio or Internet (Echolink and Allstar..no IRLP yet). I created a pair of Twitter accounts; one for the node and another for me personally. I added my node's Twitter account to the ones I follow with my personal account and created an AGI scripts that formats a curl command which posts some of the context variables passed in when Asterisk invokes the script (context name, extension, callerid, etc). I then added the AGI call into various extension stanzas in extensions.conf to test. If, for instance, I do a reverse autopatch and connect to the reverse autopatch "extension", I almost immediately get an SMS tweet on the Blackberry. Great. However, it seems that connections via the radio are not processed in any way in extensions.conf (thought [default] stanza would apply but apparently only applies to autopatch), and inbound connections from either Allstar or Echolink, while processed through [radio-secure], hang on the AGI call and don't proceed to the following call to rpt. I'm assuming this is because AGI calls generally follow an "Answer" and are only valid in the connected state. I tried "deadAGI" since it doesn't seems to be dependent on being in the answered state but it hung the same way. It's possible I've got a problem in the AGI script which is causing the hang but it certainly doesn't occur when the AGI call follows "Answer". So I'm looking for help to understand other possible ways to invoke a script that calls curl to post the tweet when a radio user makes an outbound connection to Allstar or Echolink and when an Echolink or Allstar user connects in to the radio. Ideas? Thanks and 73's Keith KF7DRV

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