- App_rpt-users - lists.keekles.org

[App_rpt] Respository and Update Script Status
by Steven Henke 24 Jan '08

24 Jan '08

Hello All, The trunk path of the app_rpt repository has been under contruction for the past couple of days. If you did an update during that time it is possible that your version of svn_rpt_update and/or your app_rpt software will not work properly. You can do the following to make sure you get the latest software properly installed: cd /usr/local/bin wget -r http://xelatec.com/asterisk/svn_rpt_update Then run svn_rpt_update. Lots of things should be improved, most notably the delay through the system. There's more work to be done but let's see how this works. If you get a problem that just can't wait, you can call me at 678-248-2979. Dial extension 215 at the IVR. 73, Steven Henke, W9SH P.S. I still want to try and have a conference call! Any time preferences?

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[App_rpt] Asterisk app_rpt Subversion Repository Temporarily Closed for Business
by Steven Henke 19 Jan '08

19 Jan '08

Gents, I'm releasing some new code, install and update scripts. In the process of getting there, the repository contents are not yet reconciled with each other. Therefore, I closed it rather than everyone getting something that is broken. It should be back up Saturday. Steve H.

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Re: [App_rpt] App_rpt Digest, Vol 23, Issue 1
by Nevada Amateur Radio Repeaters, Inc. 15 Jan '08

15 Jan '08

For the past six years a VoIP conference has been held the week end before NAB in Las Vegas. See http://www.narri.org/Annual_IRLP_Conference.html The key-note speaker has been Dave VE7LTD, the System designer for IRLP. For family reasons he is no longer available for the conference or even Dayton. It should be noted that Steven Rodgers provide a very good session on the All-Star System. Is there interest in members of this group attending and presenting at such a meeting? Kent W7AOR All-Star Node 2030 -----Original Message----- From: app_rpt-bounces(a)lists.illiana.net [mailto:app_rpt-bounces@lists.illiana.net] On Behalf Of app_rpt-request(a)lists.illiana.net Sent: Sunday, January 13, 2008 5:49 PM To: app_rpt(a)lists.illiana.net Subject: App_rpt Digest, Vol 23, Issue 1 Send App_rpt mailing list submissions to app_rpt(a)lists.illiana.net To subscribe or unsubscribe via the World Wide Web, visit http://lists.illiana.net/mailman/listinfo/app_rpt or, via email, send a message with subject or body 'help' to app_rpt-request(a)lists.illiana.net You can reach the person managing the list at app_rpt-owner(a)lists.illiana.net When replying, please edit your Subject line so it is more specific than "Re: Contents of App_rpt digest..." Today's Topics: 1. Test message - Disregard (Stephen Rodgers) 2. [Fwd: Re: USB radio interface problems] (Stephen Rodgers) 3. Asterisk app_rpt Conference Call Schedule (Steven Henke) 4. Re: Asterisk app_rpt Conference Call Schedule (n4charlielimaalpha) 5. Re: Asterisk app_rpt Conference Call Schedule (F6HQZ) 6. New Asterisk w/app_rpt Installation Package Available (Steve Henke) ---------------------------------------------------------------------- Message: 1 Date: Sun, 06 Jan 2008 11:46:42 -0800 From: Stephen Rodgers <hwstar(a)rodgers.sdcoxmail.com> Subject: [App_rpt] Test message - Disregard To: Asterisk Repeater Controler <app_rpt(a)lists.illiana.net> Message-ID: <47813022.2030301(a)rodgers.sdcoxmail.com> Content-Type: text/plain; charset=ISO-8859-1 This is a test ------------------------------ Message: 2 Date: Sun, 06 Jan 2008 16:17:20 -0800 From: Stephen Rodgers <hwstar(a)rodgers.sdcoxmail.com> Subject: [App_rpt] [Fwd: Re: USB radio interface problems] To: Asterisk Repeater Controler <app_rpt(a)lists.illiana.net> Message-ID: <47816F90.7060608(a)rodgers.sdcoxmail.com> Content-Type: text/plain; charset=ISO-8859-15 Forwarded to the list-- -------- Original Message -------- From: - Sun Jan 6 16:08:57 2008 X-Account-Key: account2 X-UIDL: <1199662935.478167577a755(a)www.kopteri.net> X-Mozilla-Status: 0011 X-Mozilla-Status2: 00000000 Return-Path: <tomi.viikari(a)i-isolutions.fi> Received: from dukecmimpi02.coxmail.com ([68.99.120.133]) by dukecmmtai01.coxmail.com (InterMail vM.6.01.06.01 201-2131-130-101-20060113) with ESMTP id <20080106234229.CXAE1976.dukecmmtai01.coxmail.com(a)dukecmimpi02.coxmail.com> for <hwstar(a)rodgers.sdcoxmail.com>; Sun, 6 Jan 2008 18:42:29 -0500 Received: from smtp.kopteri.net ([80.95.128.14]) by dukecmimpi02.coxmail.com with IMP id ZzhC1Y0200JmMgY0000000; Sun, 06 Jan 2008 18:41:15 -0500 X-Virus-Scanned: amavisd-new at kopteri.net Received: from www.kopteri.net (www.kopteri.net [80.95.128.10]) by smtp.kopteri.net (8.13.4/8.13.4/Debian-3sarge3) with ESMTP id m06NgIGp010308; Mon, 7 Jan 2008 01:42:19 +0200 Received: from www.kopteri.net (netland@localhost [127.0.0.1]) by www.kopteri.net (8.13.8/8.13.8/Debian-3) with ESMTP id m06NgIY7028727; Mon, 7 Jan 2008 01:42:18 +0200 Received: (from netland@localhost) by www.kopteri.net (8.13.8/8.13.8/Submit) id m06NgFCT028720; Mon, 7 Jan 2008 01:42:15 +0200 Received: from statip-80-95-138-16.kopteri.net (statip-80-95-138-16.kopteri.net [80.95.138.16]) by www.kopteri.net (IMP) with HTTP for <to-5069@v2_huolto>; Mon, 07 Jan 2008 01:42:15 +0200 Message-ID: <1199662935.478167577a755(a)www.kopteri.net> Date: Mon, 07 Jan 2008 01:42:15 +0200 From: Tomi Viikari <Tomi.Viikari(a)i-isolutions.fi> To: Stephen Rodgers <hwstar(a)rodgers.sdcoxmail.com>, app_rpt(a)marauder.illiana.net Cc: Steven Henke <sph(a)xelatec.com> Subject: Re: USB radio interface problems References: <004901c85086$7bb35850$0800000a@tomi> <478134A9.8060209(a)rodgers.sdcoxmail.com> In-Reply-To: <478134A9.8060209(a)rodgers.sdcoxmail.com> MIME-Version: 1.0 Content-Type: text/plain; charset=ISO-8859-15 Content-Transfer-Encoding: 8bit User-Agent: Internet Messaging Program (IMP) 3.2.1 Lainaus Stephen Rodgers <hwstar(a)rodgers.sdcoxmail.com>: > Tomi, > > Here are some things to try-- thanks for quick answer. i tested some more.... > > Steve. > > > Tomi Viikari wrote: > > Hi > > > > i have asterisk (EVB) and appr_rpt installed with usb sound/radio > interface. > > one linksys voip phone as client (sip) > > > > when i dial from sip client to asterisk/rpt in phone mode > > - there is no rx audio to sip client > > - *6 activates PTT but i can'n deactivate it (exept hanging up the phone) > > - tx audio is ok > > This is the normal way it works as you are typically use a repeater > (full duplex) for phone traffic. my configuration is simplex (duplex=0) > > -- however-- > > In the latest app_rpt.c (Version 0.91) there is an experimental simplex > autopatch which is voice-activated (VOX) from the both the autopatch and > the phone control features. > > To set it up, set duplex = 0 in rpt.conf for the node, and then place > these values in the node stanza: > > ;The following parameters are for vox operation(s) > voxtimeout=5000 ; time to let vox talk in ms > voxrecover=500 ; time to take from vox one timed out in ms > simplexpatchdelay=25 ; # of 20 ms frames to delay audio > simplexphonedelay=25 ; # of 20 ms frames to delay audio > > > For phone mode, append a v on the end of the rpt call in extensions.conf > like this: > > exten=1999,n,Rpt,1999|Pv > > This activates the VOX when in phone mode. > Vox works fine.(with no receive audio) (very useful feature to my setup) in P mode(no vox), when dialin *6 (6=cop,6) it does not activate PTT until timeout or hangup. When i enable debug in rpt, and dial first *6 from sip rpt detects *6 from sip and prints ... @@@ action cop,6 ... <<radio tx on >> and ptt activates and so on... (i think "radio tx on" is from chan_usbradio) when i dial second *6 from sip (to deactivate) it prints same @@@ action cop,6 but does not print << radio tx off >> when i dial randomly *,6,# numbers from sip in random order many times i occasionally get << radio tx off>> << radio tx on>> always in pairs so that PTT remains on. in S mode i can't activate PTT at all. dialing * from sip i get << radio tx on>> << radio tx off>> always in pairs so that PTT remains off > > > > > > some debug > > - app_rpt detects and hears dtmf commands from remote radio > > - test signal (milliwatt) from sip works fine > > Are you sure this isn't chan_usb radio reporting the DTMF tones? you are right it is chan_usb radio that reports DTMF tones > > Are you using the same CTCSS tone used during setup? i'm not using CTCSS at all > Please note that a bug has been reported on losing receive audio on some > systems with the USB FOB after running a system for a while, but in your > case it seems like the audio was never there to start with. This is not > an issue on the pciradio radio interface. Do you initially have receive > audio when you first start Asterisk? This I'd like to know. no receive audio at all. i started asterisk always fully when tested (not reload) and i have never heard receive audio > > > > - app_rpt test signal works. ( activates PTT, plays signal (in boths sip > and > > radio), and deactivates ptt) > > - hidthread detects squels (vol dn) (when manualy pressing the vol dn, > > nowired squels to radio) > > - i have tested this with modified usb-adabter and original with > headphones > > (both same result) > > - i tryed configure rpt as dupled and simplex (usbradio.conf) with no > > change > > If you are running chan_usbradio with a half-duplex radio. Set it to 0, > otherwise set it to 1. If you set up half-duplex in usbradio.conf, be > sure to use duplex mode 0 or 1 in rpt.conf as well. Both conf files need > to match each other in settings. half-duplex radio. dublex=0 on both > > > > - rpt detects dtmf's from sip > > - txtimeout works ( cut's tx after 30 sec) > > > > any ideas to try? ------------------------------ Message: 3 Date: Sun, 13 Jan 2008 13:25:49 -0500 From: "Steven Henke" <steve.w9sh(a)gmail.com> Subject: [App_rpt] Asterisk app_rpt Conference Call Schedule To: app_rpt(a)lists.illiana.net Message-ID: <e2ed85230801131025m6abeee9fo369d323c6e6e6532(a)mail.gmail.com> Content-Type: text/plain; charset="iso-8859-1" Hello Asterisk app_rpt Community, Let's talk! Through the facilities of Aretta Communications <http://aretta.com> I am organizing and hosting Asterisk app_rpt conference calls. The purpose of these calls is to let users and developers informally share information, listen to questions and concerns, make announcements and generally get to know each other. I propose to hold our first conference call this Sunday, January 13th at 6PM EDT (2300Z). From there we can agree on dates and times for future calls. I'll be hanging out on the conference bridge from time to time for anyone who wants to test their connection. Our initial agenda will consist of introductions and a brief description of each participant's existing or planned app_rpt system(s). I'll share my observations on the state of the project and my plans for contributions. Here is the SIP user information to connect to the conference call host server as an extension: Host = steveh.aretta.net User = 123 Secret = roip54 Call into extension 124 and you will be connected to the conference bridge. I hope to talk to you then. Best Regards, Steven Henke, W9SH <http://www.xelatec.com/w9sh>

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Re: [App_rpt] Asterisk app_rpt Conference Call Schedule
by n4charlielimaalpha 14 Jan '08

14 Jan '08

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[App_rpt] Asterisk app_rpt Conference Call Schedule
by Steven Henke 14 Jan '08

14 Jan '08

Hello Asterisk app_rpt Community, Let's talk! Through the facilities of Aretta Communications <http://aretta.com> I am organizing and hosting Asterisk app_rpt conference calls. The purpose of these calls is to let users and developers informally share information, listen to questions and concerns, make announcements and generally get to know each other. I propose to hold our first conference call this Sunday, January 13th at 6PM EDT (2300Z). From there we can agree on dates and times for future calls. I'll be hanging out on the conference bridge from time to time for anyone who wants to test their connection. Our initial agenda will consist of introductions and a brief description of each participant's existing or planned app_rpt system(s). I'll share my observations on the state of the project and my plans for contributions. Here is the SIP user information to connect to the conference call host server as an extension: Host = steveh.aretta.net User = 123 Secret = roip54 Call into extension 124 and you will be connected to the conference bridge. I hope to talk to you then. Best Regards, Steven Henke, W9SH <http://www.xelatec.com/w9sh>

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[App_rpt] [Fwd: Re: USB radio interface problems]
by Stephen Rodgers 07 Jan '08

07 Jan '08

Forwarded to the list-- -------- Original Message -------- From: - Sun Jan 6 16:08:57 2008 X-Account-Key: account2 X-UIDL: <1199662935.478167577a755(a)www.kopteri.net> X-Mozilla-Status: 0011 X-Mozilla-Status2: 00000000 Return-Path: <tomi.viikari(a)i-isolutions.fi> Received: from dukecmimpi02.coxmail.com ([68.99.120.133]) by dukecmmtai01.coxmail.com (InterMail vM.6.01.06.01 201-2131-130-101-20060113) with ESMTP id <20080106234229.CXAE1976.dukecmmtai01.coxmail.com(a)dukecmimpi02.coxmail.com> for <hwstar(a)rodgers.sdcoxmail.com>; Sun, 6 Jan 2008 18:42:29 -0500 Received: from smtp.kopteri.net ([80.95.128.14]) by dukecmimpi02.coxmail.com with IMP id ZzhC1Y0200JmMgY0000000; Sun, 06 Jan 2008 18:41:15 -0500 X-Virus-Scanned: amavisd-new at kopteri.net Received: from www.kopteri.net (www.kopteri.net [80.95.128.10]) by smtp.kopteri.net (8.13.4/8.13.4/Debian-3sarge3) with ESMTP id m06NgIGp010308; Mon, 7 Jan 2008 01:42:19 +0200 Received: from www.kopteri.net (netland@localhost [127.0.0.1]) by www.kopteri.net (8.13.8/8.13.8/Debian-3) with ESMTP id m06NgIY7028727; Mon, 7 Jan 2008 01:42:18 +0200 Received: (from netland@localhost) by www.kopteri.net (8.13.8/8.13.8/Submit) id m06NgFCT028720; Mon, 7 Jan 2008 01:42:15 +0200 Received: from statip-80-95-138-16.kopteri.net (statip-80-95-138-16.kopteri.net [80.95.138.16]) by www.kopteri.net (IMP) with HTTP for <to-5069@v2_huolto>; Mon, 07 Jan 2008 01:42:15 +0200 Message-ID: <1199662935.478167577a755(a)www.kopteri.net> Date: Mon, 07 Jan 2008 01:42:15 +0200 From: Tomi Viikari <Tomi.Viikari(a)i-isolutions.fi> To: Stephen Rodgers <hwstar(a)rodgers.sdcoxmail.com>, app_rpt(a)marauder.illiana.net Cc: Steven Henke <sph(a)xelatec.com> Subject: Re: USB radio interface problems References: <004901c85086$7bb35850$0800000a@tomi> <478134A9.8060209(a)rodgers.sdcoxmail.com> In-Reply-To: <478134A9.8060209(a)rodgers.sdcoxmail.com> MIME-Version: 1.0 Content-Type: text/plain; charset=ISO-8859-15 Content-Transfer-Encoding: 8bit User-Agent: Internet Messaging Program (IMP) 3.2.1 Lainaus Stephen Rodgers <hwstar(a)rodgers.sdcoxmail.com>: > Tomi, > > Here are some things to try-- thanks for quick answer. i tested some more.... > > Steve. > > > Tomi Viikari wrote: > > Hi > > > > i have asterisk (EVB) and appr_rpt installed with usb sound/radio > interface. > > one linksys voip phone as client (sip) > > > > when i dial from sip client to asterisk/rpt in phone mode > > - there is no rx audio to sip client > > - *6 activates PTT but i can'n deactivate it (exept hanging up the phone) > > - tx audio is ok > > This is the normal way it works as you are typically use a repeater > (full duplex) for phone traffic. my configuration is simplex (duplex=0) > > -- however-- > > In the latest app_rpt.c (Version 0.91) there is an experimental simplex > autopatch which is voice-activated (VOX) from the both the autopatch and > the phone control features. > > To set it up, set duplex = 0 in rpt.conf for the node, and then place > these values in the node stanza: > > ;The following parameters are for vox operation(s) > voxtimeout=5000 ; time to let vox talk in ms > voxrecover=500 ; time to take from vox one timed out in ms > simplexpatchdelay=25 ; # of 20 ms frames to delay audio > simplexphonedelay=25 ; # of 20 ms frames to delay audio > > > For phone mode, append a v on the end of the rpt call in extensions.conf > like this: > > exten=1999,n,Rpt,1999|Pv > > This activates the VOX when in phone mode. > Vox works fine.(with no receive audio) (very useful feature to my setup) in P mode(no vox), when dialin *6 (6=cop,6) it does not activate PTT until timeout or hangup. When i enable debug in rpt, and dial first *6 from sip rpt detects *6 from sip and prints ... @@@ action cop,6 ... <<radio tx on >> and ptt activates and so on... (i think "radio tx on" is from chan_usbradio) when i dial second *6 from sip (to deactivate) it prints same @@@ action cop,6 but does not print << radio tx off >> when i dial randomly *,6,# numbers from sip in random order many times i occasionally get << radio tx off>> << radio tx on>> always in pairs so that PTT remains on. in S mode i can't activate PTT at all. dialing * from sip i get << radio tx on>> << radio tx off>> always in pairs so that PTT remains off > > > > > > some debug > > - app_rpt detects and hears dtmf commands from remote radio > > - test signal (milliwatt) from sip works fine > > Are you sure this isn't chan_usb radio reporting the DTMF tones? you are right it is chan_usb radio that reports DTMF tones > > Are you using the same CTCSS tone used during setup? i'm not using CTCSS at all > Please note that a bug has been reported on losing receive audio on some > systems with the USB FOB after running a system for a while, but in your > case it seems like the audio was never there to start with. This is not > an issue on the pciradio radio interface. Do you initially have receive > audio when you first start Asterisk? This I'd like to know. no receive audio at all. i started asterisk always fully when tested (not reload) and i have never heard receive audio > > > > - app_rpt test signal works. ( activates PTT, plays signal (in boths sip > and > > radio), and deactivates ptt) > > - hidthread detects squels (vol dn) (when manualy pressing the vol dn, > > nowired squels to radio) > > - i have tested this with modified usb-adabter and original with > headphones > > (both same result) > > - i tryed configure rpt as dupled and simplex (usbradio.conf) with no > > change > > If you are running chan_usbradio with a half-duplex radio. Set it to 0, > otherwise set it to 1. If you set up half-duplex in usbradio.conf, be > sure to use duplex mode 0 or 1 in rpt.conf as well. Both conf files need > to match each other in settings. half-duplex radio. dublex=0 on both > > > > - rpt detects dtmf's from sip > > - txtimeout works ( cut's tx after 30 sec) > > > > any ideas to try?

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[App_rpt] Test message - Disregard
by Stephen Rodgers 06 Jan '08

06 Jan '08

This is a test

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[App_rpt] Hello
by Don Russell 30 Dec '07

30 Dec '07

New to the list and wanted to say hello. Working on building a node with the usb fob. Located in Rockford, IL. 73's Don _________________________________________________________________ Get the power of Windows + Web with the new Windows Live. http://www.windowslive.com?ocid=TXT_TAGHM_Wave2_powerofwindows_122007

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[App_rpt] Node 2047 up, running usb sound fob and a question on the node operation
by Will 29 Dec '07

29 Dec '07

Steve, and the group, I now have node 2047 up. This is a Radius M10 2m link radio. Question: Once I connect node 2047 to the Repeater node, 2059, I get the echo sound and some time lose control over voice and dtmf signals. Both nodes run off the same computer. 933Mhz 512mb of memory. Any thoughts will be appreciated. Oh and running both by the usb sound fob. Will / W4WWM

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[App_rpt] EVB 0.011
by David Powell-Williams 06 Dec '07

06 Dec '07

Steve: First, thank you for your quick reply. I did another install from scratch using your link to get the bash script evbphase2. I believe this was the same version that I had used initially. However, I received many more Errors, Warnings and Notices than before, including all the ones from my first attempt. I have tried this installation on two computers but with exactly the same results: Everything is fine and Asterisk runs until I execute evbphase2. After it finishes, Asterisk starts but immediately exits with error return 1. A number of the Warnings are really puzzling, such as looking for the sound module 'format_au.so' which clearly does not exist on disk. Is it possible that you had some residual files on your system when you tested evbphase2? I can send you the final lines from /var/log/asterisk/full or the death messages from Asterisk if you would like to see them. Do you see any advantage in trying EVB 010? Looks like I may have to return to compiling everything from source onto Debian :) I would certainly appreciate any further thoughts. 73, David VE7MQ.

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