From billreid at shaw.ca Mon Mar 2 19:00:52 2009 From: billreid at shaw.ca (Bill Reid) Date: Mon, 02 Mar 2009 19:00:52 -0600 Subject: [*] Update on Tues meeting Message-ID: <49AC8144.2080905@shaw.ca> Ron has just found out that he will be traveling to Ottawa on Tues evening so he can not make the meeting. I will share some of his ideas behind his Wifi access proposal. c u tomorrow, Bill From john at johnlange.ca Tue Mar 3 09:36:19 2009 From: john at johnlange.ca (John Lange) Date: Tue, 03 Mar 2009 09:36:19 -0600 Subject: [*] Update on Tues meeting In-Reply-To: <49AC8144.2080905@shaw.ca> References: <49AC8144.2080905@shaw.ca> Message-ID: <1236094579.5408.7.camel@linux-2sym> We can conference him in if he wants ;) -- John Lange http://www.johnlange.ca On Mon, 2009-03-02 at 19:00 -0600, Bill Reid wrote: > Ron has just found out that he will be traveling to Ottawa on Tues evening so he > can not make the meeting. > > I will share some of his ideas behind his Wifi access proposal. > > c u tomorrow, > Bill > _______________________________________________ > Asterisk mailing list > Asterisk at muug.mb.ca > http://www.muug.mb.ca/mailman/listinfo/asterisk > From billreid at shaw.ca Thu Mar 5 21:48:28 2009 From: billreid at shaw.ca (Bill Reid) Date: Thu, 05 Mar 2009 21:48:28 -0600 Subject: [*] Skype For Asterisk Update Message-ID: <49B09D0C.3060601@shaw.ca> I was wondering what was happening with the Skype channel. It has been very quiet but I see that Digium issued an update a couple of weeks ago. http://blogs.digium.com/2009/02/23/skype-for-asterisk-update/ -- Bill From john at johnlange.ca Thu Mar 12 18:50:09 2009 From: john at johnlange.ca (John Lange) Date: Thu, 12 Mar 2009 18:50:09 -0500 Subject: [*] One Number to Ring All Your Phones: Google Resurrects GrandCentral Service Message-ID: <1236901809.5353.3.camel@linux-2sym> "Now that Google has meshed GrandCentral services into its massive technology infrastructure the company has added a number of other useful features. For one, you can get automated transcripts of voicemails and view them on the Web. The transcripts aren't perfect, but more than good enough to get the gist at a glance." http://www.businessweek.com/the_thread/techbeat/archives/2009/03/post_14.html I'd like to see someone hack that up on Asterisk's voice-mail-to-email. Send the email with speech recognition plus the attachment. All the pieces of the puzzle are there, they just need to be fitted together. Idea's anyone? -- John Lange http://www.johnlange.ca From trapper at foxlakecreenation.com Thu Mar 12 21:11:31 2009 From: trapper at foxlakecreenation.com (Travis Harper) Date: Thu, 12 Mar 2009 21:11:31 -0500 Subject: [*] One Number to Ring All Your Phones: Google Resurrects GrandCentral Service In-Reply-To: <1236901809.5353.3.camel@linux-2sym> References: <1236901809.5353.3.camel@linux-2sym> Message-ID: <365DDD54-3798-4C3D-967C-DF2F565A4385@foxlakecreenation.com> I have been using Google Grandcentral for sometime now. Have not had any luck with asterisk integration. I have a couple on numbers in New York City, that I use in GGC. When someone calls that number, my Cell phone, Home Phone ( asterisk ), and My inlaws house all ring at the same time. What I would like see is Google offering sip trunking. Any ideas if that will available or even possible? Travis On 12-Mar-09, at 6:50 PM, John Lange wrote: > "Now that Google has meshed GrandCentral services into its massive > technology infrastructure the company has added a number of other > useful > features. For one, you can get automated transcripts of voicemails and > view them on the Web. The transcripts aren't perfect, but more than > good > enough to get the gist at a glance." > > http://www.businessweek.com/the_thread/techbeat/archives/2009/03/ > post_14.html > > I'd like to see someone hack that up on Asterisk's voice-mail-to- > email. > Send the email with speech recognition plus the attachment. > > All the pieces of the puzzle are there, they just need to be fitted > together. > > Idea's anyone? > > -- > John Lange > http://www.johnlange.ca > > > _______________________________________________ > Asterisk mailing list > Asterisk at muug.mb.ca > http://www.muug.mb.ca/mailman/listinfo/asterisk From john at johnlange.ca Wed Mar 18 16:32:46 2009 From: john at johnlange.ca (John Lange) Date: Wed, 18 Mar 2009 16:32:46 -0500 Subject: [*] The latest on Skype for Asterisk Message-ID: <1237411966.5083.125.camel@linux-2sym> http://ecommconf.com/blog/2009/03/mark-spencer-transcript.html -- John Lange http://www.johnlange.ca From swalberg at gmail.com Thu Mar 19 07:59:56 2009 From: swalberg at gmail.com (Sean Walberg) Date: Thu, 19 Mar 2009 07:59:56 -0500 Subject: [*] Google Measurement Lab, Wireshark, Sharkfest 09 Message-ID: There was talk earlier about the Google Measurement Lab which is the suite of online tools to see, among other things, if you're being throttled. The Wireshark conference, Sharkfest 09, is now featuring a session on the tool set. The conference runs from June 15 - 18th. I'll be at the conference again this year. Last year was an awesome time, I learned an incredible amount about Wireshark (and I use it on a daily basis). It's at Standford this year. Sean -- Sean Walberg http://ertw.com/ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://www.muug.mb.ca/pipermail/asterisk/attachments/20090319/86df9ab6/attachment.html From john at johnlange.ca Tue Mar 24 14:09:01 2009 From: john at johnlange.ca (John Lange) Date: Tue, 24 Mar 2009 14:09:01 -0500 Subject: [*] Generating test audio Message-ID: <1237921749.10614.55.camel@linux-2sym> I'm looking for suggestions on how to generate test audio from my laptop that can be piped through Asterisk and ultimately to an endpoint. It would be best if the audio were a continuous tone but music or something else would be acceptable also. I thought maybe I could use twinkle and set it's input audio device to be the output of say a streaming radio site or something but I haven't been able to get that to work. Any suggestions? -- John Lange http://www.johnlange.ca From johannes.vanderknyff at gmail.com Tue Mar 24 14:21:15 2009 From: johannes.vanderknyff at gmail.com (Johannes Vanderknyff) Date: Tue, 24 Mar 2009 15:21:15 -0400 Subject: [*] [on-asterisk] Generating test audio In-Reply-To: <1237921749.10614.55.camel@linux-2sym> References: <1237921749.10614.55.camel@linux-2sym> Message-ID: Just thinking out loud, but I've worked with TV-capture cards and have used short little cables that will connect the line-out to the line-in...... maybe that would work?.... Johannes On Tue, Mar 24, 2009 at 3:09 PM, John Lange wrote: > I'm looking for suggestions on how to generate test audio from my laptop > that can be piped through Asterisk and ultimately to an endpoint. > > It would be best if the audio were a continuous tone but music or > something else would be acceptable also. > > I thought maybe I could use twinkle and set it's input audio device to > be the output of say a streaming radio site or something but I haven't > been able to get that to work. > > Any suggestions? > > -- > John Lange > http://www.johnlange.ca > > > > --------------------------------------------------------------------- > To unsubscribe, e-mail: asterisk-unsubscribe at uc.org > For additional commands, e-mail: asterisk-help at uc.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://www.muug.mb.ca/pipermail/asterisk/attachments/20090324/ae2ce475/attachment.html From john at johnlange.ca Tue Mar 24 14:27:42 2009 From: john at johnlange.ca (John Lange) Date: Tue, 24 Mar 2009 14:27:42 -0500 Subject: [*] Generating test audio In-Reply-To: References: <1237921749.10614.55.camel@linux-2sym> Message-ID: <1237922863.10614.87.camel@linux-2sym> On Tue, 2009-03-24 at 14:24 -0500, Sean Walberg wrote: > VLC will stream files to an unicast/mcast address. You could make a > wav file with the tone, and tell VLC to loop. You'll have to connect the dots for me; how do I get that output into Twinkle or alternatively into a SIP/RTP stream? John > > Sean > > On Tue, Mar 24, 2009 at 2:09 PM, John Lange wrote: > I'm looking for suggestions on how to generate test audio from > my laptop > that can be piped through Asterisk and ultimately to an > endpoint. > > It would be best if the audio were a continuous tone but music > or > something else would be acceptable also. > > I thought maybe I could use twinkle and set it's input audio > device to > be the output of say a streaming radio site or something but I > haven't > been able to get that to work. > > Any suggestions? > > -- > John Lange > http://www.johnlange.ca > > > _______________________________________________ > Asterisk mailing list > Asterisk at muug.mb.ca > http://www.muug.mb.ca/mailman/listinfo/asterisk > > > > -- > Sean Walberg http://ertw.com/ From swalberg at gmail.com Tue Mar 24 14:24:21 2009 From: swalberg at gmail.com (Sean Walberg) Date: Tue, 24 Mar 2009 14:24:21 -0500 Subject: [*] Generating test audio In-Reply-To: <1237921749.10614.55.camel@linux-2sym> References: <1237921749.10614.55.camel@linux-2sym> Message-ID: VLC will stream files to an unicast/mcast address. You could make a wav file with the tone, and tell VLC to loop. Sean On Tue, Mar 24, 2009 at 2:09 PM, John Lange wrote: > I'm looking for suggestions on how to generate test audio from my laptop > that can be piped through Asterisk and ultimately to an endpoint. > > It would be best if the audio were a continuous tone but music or > something else would be acceptable also. > > I thought maybe I could use twinkle and set it's input audio device to > be the output of say a streaming radio site or something but I haven't > been able to get that to work. > > Any suggestions? > > -- > John Lange > http://www.johnlange.ca > > > _______________________________________________ > Asterisk mailing list > Asterisk at muug.mb.ca > http://www.muug.mb.ca/mailman/listinfo/asterisk > -- Sean Walberg http://ertw.com/ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://www.muug.mb.ca/pipermail/asterisk/attachments/20090324/f3a0524c/attachment.html From billreid at shaw.ca Tue Mar 24 14:30:36 2009 From: billreid at shaw.ca (Bill Reid) Date: Tue, 24 Mar 2009 14:30:36 -0500 Subject: [*] Generating test audio In-Reply-To: <1237921749.10614.55.camel@linux-2sym> References: <1237921749.10614.55.camel@linux-2sym> Message-ID: <49C934DC.4050508@shaw.ca> Have you tried using sox to convert an mp3 to gsm or ulaw? John Lange wrote: > I'm looking for suggestions on how to generate test audio from my laptop > that can be piped through Asterisk and ultimately to an endpoint. > > It would be best if the audio were a continuous tone but music or > something else would be acceptable also. > > I thought maybe I could use twinkle and set it's input audio device to > be the output of say a streaming radio site or something but I haven't > been able to get that to work. > > Any suggestions? > From billreid at shaw.ca Tue Mar 24 14:32:01 2009 From: billreid at shaw.ca (Bill Reid) Date: Tue, 24 Mar 2009 14:32:01 -0500 Subject: [*] Generating test audio In-Reply-To: <1237921749.10614.55.camel@linux-2sym> References: <1237921749.10614.55.camel@linux-2sym> Message-ID: <49C93531.7000602@shaw.ca> Have you tried using sox to convert an mp3 to gsm or ulaw? John Lange wrote: > I'm looking for suggestions on how to generate test audio from my laptop > that can be piped through Asterisk and ultimately to an endpoint. > > It would be best if the audio were a continuous tone but music or > something else would be acceptable also. > > I thought maybe I could use twinkle and set it's input audio device to > be the output of say a streaming radio site or something but I haven't > been able to get that to work. > > Any suggestions? > From swalberg at gmail.com Tue Mar 24 14:39:19 2009 From: swalberg at gmail.com (Sean Walberg) Date: Tue, 24 Mar 2009 14:39:19 -0500 Subject: [*] Generating test audio In-Reply-To: <1237922863.10614.87.camel@linux-2sym> References: <1237921749.10614.55.camel@linux-2sym> <1237922863.10614.87.camel@linux-2sym> Message-ID: On Tue, Mar 24, 2009 at 2:27 PM, John Lange wrote: > On Tue, 2009-03-24 at 14:24 -0500, Sean Walberg wrote: > > VLC will stream files to an unicast/mcast address. You could make a > > wav file with the tone, and tell VLC to loop. > > You'll have to connect the dots for me; how do I get that output into > Twinkle or alternatively into a SIP/RTP stream? One of the streaming output options is RTP. I'm not quite sure what you're trying to do, so the signalling might be the problem. Sean -- Sean Walberg http://ertw.com/ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://www.muug.mb.ca/pipermail/asterisk/attachments/20090324/cdd2f8ce/attachment.html From billreid at shaw.ca Tue Mar 24 14:42:45 2009 From: billreid at shaw.ca (Bill Reid) Date: Tue, 24 Mar 2009 14:42:45 -0500 Subject: [*] Generating test audio In-Reply-To: <49C93531.7000602@shaw.ca> References: <1237921749.10614.55.camel@linux-2sym> <49C93531.7000602@shaw.ca> Message-ID: <49C937B5.5030807@shaw.ca> I did not understand completely what was your problem. I think SIPp is what you what. I did give a demo of it a while back. http://sipp.sourceforge.net/ "SIPp is a performance testing tool for the SIP protocol. It includes a few basic SipStone user agent scenarios (UAC and UAS) and establishes and releases multiple calls with the INVITE and BYE methods. It can also reads XML scenario files describing any performance testing configuration. It features the dynamic display of statistics about running tests (call rate, round trip delay, and message statistics), periodic CSV statistics dumps, TCP and UDP over multiple sockets or multiplexed with retransmission management, regular expressions and variables in scenario files, and dynamically adjustable call rates." - Bill Bill Reid wrote: > Have you tried using sox to convert an mp3 to gsm or ulaw? > > John Lange wrote: >> I'm looking for suggestions on how to generate test audio from my laptop >> that can be piped through Asterisk and ultimately to an endpoint. >> >> It would be best if the audio were a continuous tone but music or >> something else would be acceptable also. >> >> I thought maybe I could use twinkle and set it's input audio device to >> be the output of say a streaming radio site or something but I haven't >> been able to get that to work. >> >> Any suggestions? >> > > > _______________________________________________ > Asterisk mailing list > Asterisk at muug.mb.ca > http://www.muug.mb.ca/mailman/listinfo/asterisk > From john at johnlange.ca Tue Mar 24 14:56:14 2009 From: john at johnlange.ca (John Lange) Date: Tue, 24 Mar 2009 14:56:14 -0500 Subject: [*] Generating test audio In-Reply-To: <49C937B5.5030807@shaw.ca> References: <1237921749.10614.55.camel@linux-2sym> <49C93531.7000602@shaw.ca> <49C937B5.5030807@shaw.ca> Message-ID: <1237924575.10614.121.camel@linux-2sym> On Tue, 2009-03-24 at 14:42 -0500, Bill Reid wrote: > I did not understand completely what was your problem. I think > SIPp is what you what. I did give a demo of it a while back. Yes, but after spending hours on it and not getting it working I thought "there must be an easier way". With SIPp I can setup the test call and play the audio but the audio does not make it to the far end. My guess is Asterisk doesn't like the handshake or the one-way audio, or SIPp is sending the stream to the wrong destination port? There are so many things that could be the problem and all the while I'm trouble shooting SIPp and not even working on what I'm trying to test. - John Lange http://www.johnlange.ca From john at johnlange.ca Tue Mar 24 15:10:23 2009 From: john at johnlange.ca (John Lange) Date: Tue, 24 Mar 2009 15:10:23 -0500 Subject: [*] Generating test audio In-Reply-To: References: <1237921749.10614.55.camel@linux-2sym> <1237922863.10614.87.camel@linux-2sym> Message-ID: <1237925423.10614.136.camel@linux-2sym> On Tue, 2009-03-24 at 14:39 -0500, Sean Walberg wrote: > On Tue, Mar 24, 2009 at 2:27 PM, John Lange wrote: > On Tue, 2009-03-24 at 14:24 -0500, Sean Walberg wrote: > > VLC will stream files to an unicast/mcast address. You > could make a > > wav file with the tone, and tell VLC to loop. > > > You'll have to connect the dots for me; how do I get that > output into > Twinkle or alternatively into a SIP/RTP stream? > > One of the streaming output options is RTP. I'm not quite sure what > you're trying to do, so the signalling might be the problem. To be more specific; I'm trying to generate a test signal on my laptop that I can then route through Asterisk and listen to on some other device, for example my cell phone. In this way I can tweak the QOS settings on a firewall and hear results instantly. At the moment the only way I can think of to do this is to install a full-blow instance of Asterisk, setup sip-trunking between it and the destination Asterisk and then make up some test-call files and drop them into /var/spool/asterisk/outgoing. Just seems like overkill. John From swalberg at gmail.com Tue Mar 24 15:20:32 2009 From: swalberg at gmail.com (Sean Walberg) Date: Tue, 24 Mar 2009 15:20:32 -0500 Subject: [*] Generating test audio In-Reply-To: <1237925423.10614.136.camel@linux-2sym> References: <1237921749.10614.55.camel@linux-2sym> <1237922863.10614.87.camel@linux-2sym> <1237925423.10614.136.camel@linux-2sym> Message-ID: On Tue, Mar 24, 2009 at 3:10 PM, John Lange wrote: > you're trying to do, so the signalling might be the problem. > > To be more specific; I'm trying to generate a test signal on my laptop > that I can then route through Asterisk and listen to on some other > device, for example my cell phone. Can you use a multicast source for MoH? That's \what I'm trying to do for Cisco Call Manager. The system tells the phone or gateway to listen to a multicast stream that's just RTP. You'd think you could do the same in Asterisk Sean -- Sean Walberg http://ertw.com/ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://www.muug.mb.ca/pipermail/asterisk/attachments/20090324/b99634eb/attachment.html From cliff at jazinga.com Tue Mar 24 14:52:10 2009 From: cliff at jazinga.com (Cliff Flood) Date: Tue, 24 Mar 2009 15:52:10 -0400 Subject: [*] [on-asterisk] Generating test audio In-Reply-To: <1237921749.10614.55.camel@linux-2sym> References: <1237921749.10614.55.camel@linux-2sym> Message-ID: <49C939EA.100@jazinga.com> John Lange wrote: > I'm looking for suggestions on how to generate test audio from my laptop > that can be piped through Asterisk and ultimately to an endpoint. > > It would be best if the audio were a continuous tone but music or > something else would be acceptable also. > > I thought maybe I could use twinkle and set it's input audio device to > be the output of say a streaming radio site or something but I haven't > been able to get that to work. > > Any suggestions? Hi John, consider using a milliwatt test for this: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Milliwatt Regards, -- Cliff Flood Systems Administrator Jazinga Inc. +1 416 548 4755 x106 From billreid at shaw.ca Tue Mar 24 15:26:33 2009 From: billreid at shaw.ca (Bill Reid) Date: Tue, 24 Mar 2009 15:26:33 -0500 Subject: [*] Generating test audio In-Reply-To: <1237925423.10614.136.camel@linux-2sym> References: <1237921749.10614.55.camel@linux-2sym> <1237922863.10614.87.camel@linux-2sym> <1237925423.10614.136.camel@linux-2sym> Message-ID: <49C941F9.2080300@shaw.ca> John Lange wrote: > > > In this way I can tweak the QOS settings on a firewall and hear results > instantly. > You could run two sipp UA on your laptop. The one caveat is I am not sure that you will be able to hear the received audio. -- Bill From ve4drk at gmail.com Tue Mar 24 15:28:30 2009 From: ve4drk at gmail.com (Dan Keizer) Date: Tue, 24 Mar 2009 15:28:30 -0500 Subject: [*] Generating test audio In-Reply-To: <1237925423.10614.136.camel@linux-2sym> References: <1237921749.10614.55.camel@linux-2sym> <1237922863.10614.87.camel@linux-2sym> <1237925423.10614.136.camel@linux-2sym> Message-ID: I used to run a program that would emit a specific tone (1200 Hz etc) that one could use to tune an amateur radio based AFSK modem ... it was a PC program too ... useful? Dan. On Tue, Mar 24, 2009 at 3:10 PM, John Lange wrote: > On Tue, 2009-03-24 at 14:39 -0500, Sean Walberg wrote: >> On Tue, Mar 24, 2009 at 2:27 PM, John Lange wrote: >> ? ? ? ? On Tue, 2009-03-24 at 14:24 -0500, Sean Walberg wrote: >> ? ? ? ? > VLC will stream files to an unicast/mcast address. ?You >> ? ? ? ? could make a >> ? ? ? ? > wav file with the tone, and tell VLC to loop. >> >> >> ? ? ? ? You'll have to connect the dots for me; how do I get that >> ? ? ? ? output into >> ? ? ? ? Twinkle or alternatively into a SIP/RTP stream? >> >> One of the streaming output options is RTP. I'm not quite sure what >> you're trying to do, so the signalling might be the problem. > > To be more specific; I'm trying to generate a test signal on my laptop > that I can then route through Asterisk and listen to on some other > device, for example my cell phone. > > In this way I can tweak the QOS settings on a firewall and hear results > instantly. > > At the moment the only way I can think of to do this is to install a > full-blow instance of Asterisk, setup sip-trunking between it and the > destination Asterisk and then make up some test-call files and drop them > into /var/spool/asterisk/outgoing. > > Just seems like overkill. > > John > > > > _______________________________________________ > Asterisk mailing list > Asterisk at muug.mb.ca > http://www.muug.mb.ca/mailman/listinfo/asterisk > From ve4drk at gmail.com Tue Mar 24 15:30:49 2009 From: ve4drk at gmail.com (Dan Keizer) Date: Tue, 24 Mar 2009 15:30:49 -0500 Subject: [*] Generating test audio In-Reply-To: References: <1237921749.10614.55.camel@linux-2sym> <1237922863.10614.87.camel@linux-2sym> <1237925423.10614.136.camel@linux-2sym> Message-ID: On that note .. check the following link: http://www.klm-tech.com/technicothica/xr.html At the bottom of the page are references to programs (dos and windows) for generation programs. Dan. On Tue, Mar 24, 2009 at 3:28 PM, Dan Keizer wrote: > I used to run a program that would emit a specific tone (1200 Hz etc) > that one could use to tune an amateur radio based AFSK modem ... it > was a PC program too ... ?useful? > > Dan. > > On Tue, Mar 24, 2009 at 3:10 PM, John Lange wrote: >> On Tue, 2009-03-24 at 14:39 -0500, Sean Walberg wrote: >>> On Tue, Mar 24, 2009 at 2:27 PM, John Lange wrote: >>> ? ? ? ? On Tue, 2009-03-24 at 14:24 -0500, Sean Walberg wrote: >>> ? ? ? ? > VLC will stream files to an unicast/mcast address. ?You >>> ? ? ? ? could make a >>> ? ? ? ? > wav file with the tone, and tell VLC to loop. >>> >>> >>> ? ? ? ? You'll have to connect the dots for me; how do I get that >>> ? ? ? ? output into >>> ? ? ? ? Twinkle or alternatively into a SIP/RTP stream? >>> >>> One of the streaming output options is RTP. I'm not quite sure what >>> you're trying to do, so the signalling might be the problem. >> >> To be more specific; I'm trying to generate a test signal on my laptop >> that I can then route through Asterisk and listen to on some other >> device, for example my cell phone. >> >> In this way I can tweak the QOS settings on a firewall and hear results >> instantly. >> >> At the moment the only way I can think of to do this is to install a >> full-blow instance of Asterisk, setup sip-trunking between it and the >> destination Asterisk and then make up some test-call files and drop them >> into /var/spool/asterisk/outgoing. >> >> Just seems like overkill. >> >> John >> >> >> >> _______________________________________________ >> Asterisk mailing list >> Asterisk at muug.mb.ca >> http://www.muug.mb.ca/mailman/listinfo/asterisk >> > From john at johnlange.ca Tue Mar 24 15:55:26 2009 From: john at johnlange.ca (John Lange) Date: Tue, 24 Mar 2009 15:55:26 -0500 Subject: [*] Generating test audio In-Reply-To: References: <1237921749.10614.55.camel@linux-2sym> <1237922863.10614.87.camel@linux-2sym> <1237925423.10614.136.camel@linux-2sym> Message-ID: <1237928126.10614.161.camel@linux-2sym> Just to be clear, generating the source audio is not the problem. Piping it through a SIP user-agent to Asterisk is... In any case, thanks for all the suggestions. Ultimately it just turns out to be much easier to run Asterisk on my laptop and initiate the call that way. Here is the complete solution: ;sip.conf [voip1] type=friend host=testhost.com nat=yes context=local_pstn dtmfmode=rfc2833 canreinvite=no qualify=no disallow=all allow=ulaw ; extensions.conf [default] exten => 500,1,Playback(followme/pls-hold-while-try) exten => 500,2,Milliwatt() ; testcall file: Channel: SIP/204xxxxxxx at voip1 MaxRetries: 0 RetryTime: 60 WaitTime: 30 Context: default Extension: 500 ---- Then copy the test file into the outgoing directory: cp testcall /var/spool/asterisk/outgoing/ - John Lange http://www.johnlange.ca From jim.vanmeggelen at coretel.ca Tue Mar 24 20:19:43 2009 From: jim.vanmeggelen at coretel.ca (Jim Van Meggelen) Date: Tue, 24 Mar 2009 20:19:43 -0500 (CDT) Subject: [*] Generating test audio In-Reply-To: <1237921749.10614.55.camel@linux-2sym> Message-ID: Sorry for piping in here late, but I think that YATE would be able to do that for you fairly simply. It compiles easily on Windows or Linux, and has both a call and tone generator built in. Jim Jim Van Meggelen +1-877-CORETEL (Canada) +1-866-644-7729 (USA) +1-416-425-6111 x6001 jim.vanmeggelen at coretel.ca http://www.coretel.ca http://www.iconverged.com http://downloads.oreilly.com/books/9780596510480.pdf ----- Original Message ----- From: "John Lange" To: "Asterisk Users Group" , "Asterisk SIG" Sent: Tuesday, 24 March, 2009 15:09:01 GMT -05:00 US/Canada Eastern Subject: [*] Generating test audio I'm looking for suggestions on how to generate test audio from my laptop that can be piped through Asterisk and ultimately to an endpoint. It would be best if the audio were a continuous tone but music or something else would be acceptable also. I thought maybe I could use twinkle and set it's input audio device to be the output of say a streaming radio site or something but I haven't been able to get that to work. Any suggestions? -- John Lange http://www.johnlange.ca _______________________________________________ Asterisk mailing list Asterisk at muug.mb.ca http://www.muug.mb.ca/mailman/listinfo/asterisk From john at johnlange.ca Tue Mar 31 13:16:32 2009 From: john at johnlange.ca (John Lange) Date: Tue, 31 Mar 2009 13:16:32 -0500 Subject: [*] =?utf-8?q?CRTC=E2=80=99s_online_consultation_on_Internet_traf?= =?utf-8?q?fic_management?= Message-ID: <1238523392.5362.12.camel@linux-2sym> The CRTC has setup a web site to consult the public about throttling. There is even a youtube video from the CRTC... I know you're thinking "the CRTC knows about YouTube?!", but it's for real: http://isppractices.econsultation.ca/ There is even a place to post comments. Press Release: http://crtc.gc.ca/eng/news/releases/2009/r090331.htm If you haven't already made your thoughts known to the CRTC, now is your chance. -- John Lange http://www.johnlange.ca