We are in the process of preparing to move our Asterisk server to a Digital T1 interface card instead of a analog card (via an Adtran which is now connected to the T1). I did a preliminary test the other day and hooked the T1 line up to the T1 card, bypassing the Adtran. This worked rather well I must say. The two issues I ran into are: 1) Caller ID is not working even though I enabled it. I simply do not see _anything_ related to caller ID going on. (not major, I am not even sure the phone company has it setup properly so I need to talk to them first, Verizon) 2) Asterisk is not detecting the far end hangup. Through the Adtran it does, but direct digital it does not. I bridged an incoming call to an analog phone and listened as I hung up the far end (cell-call). I hear a audible "click", silence, and then after maybe a half second I hear dialtone. I tried turning on hanguponpolarity switch, tried turning it off, tried turning callprogress on and off, still does not detect the hangup. Am I missing something obvious in Asterisk (this is my first digital hookup)? I read somewhere that Asterisk is already suppose to detect dialtone to know that the far-end hungup. Do I need to call my phone company and get details on exactly how they are triggering the hangup, though I would think with digital it "just happens"). Daniel
Digital ISDN used Q931 messages. You should get a disconnect message from telco on the d-channel 23. -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Daniel Hazelbaker Sent: Monday, July 07, 2008 4:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] US T1 Hangup Detection We are in the process of preparing to move our Asterisk server to a Digital T1 interface card instead of a analog card (via an Adtran which is now connected to the T1). I did a preliminary test the other day and hooked the T1 line up to the T1 card, bypassing the Adtran. This worked rather well I must say. The two issues I ran into are: 1) Caller ID is not working even though I enabled it. I simply do not see _anything_ related to caller ID going on. (not major, I am not even sure the phone company has it setup properly so I need to talk to them first, Verizon) 2) Asterisk is not detecting the far end hangup. Through the Adtran it does, but direct digital it does not. I bridged an incoming call to an analog phone and listened as I hung up the far end (cell-call). I hear a audible "click", silence, and then after maybe a half second I hear dialtone. I tried turning on hanguponpolarity switch, tried turning it off, tried turning callprogress on and off, still does not detect the hangup. Am I missing something obvious in Asterisk (this is my first digital hookup)? I read somewhere that Asterisk is already suppose to detect dialtone to know that the far-end hungup. Do I need to call my phone company and get details on exactly how they are triggering the hangup, though I would think with digital it "just happens"). Daniel _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ----------------------------------------- Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you.
Daniel Hazelbaker wrote:> We are in the process of preparing to move our Asterisk server to a > Digital T1 interface card instead of a analog card (via an Adtran > which is now connected to the T1). I did a preliminary test the other >A T1 or a PRI? Just make sure we're on the same page. Also, show us your zaptel and zapata.conf Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety."
> Date: Mon, 7 Jul 2008 16:48:00 -0400 > From: "Jason Aarons \(US\)" <jason.aarons at us.didata.com> > > Digital ISDN used Q931 messages. You should get a disconnect message > from telco on the d-channel 23.I am pretty sure it is a T1 and not a PRI. I did try configuring it as a PRI and it started spewing all kinds of errors and completely stopped working.> Date: Mon, 07 Jul 2008 16:55:27 -0400 > From: Doug Lytle <support at drdos.info> > > Daniel Hazelbaker wrote: >> We are in the process of preparing to move our Asterisk server to a >> Digital T1 interface card instead of a analog card (via an Adtran >> which is now connected to the T1). I did a preliminary test the >> other >> > > A T1 or a PRI? Just make sure we're on the same page. > Also, show us your zaptel and zapata.confAgain, I am pretty sure T1. It is a Verizon "Flex-Grow" package, which they list as expandable up to 24 voice channels. That and I tried configuring as a PRI and it harfed. The Adtran box we use now is configured as: Timing Mode Network Format ESF Line Code B8ZS Equalization 0 dB CSU Lpbk Enable Rx Sensitivity Auto Right now with Asterisk "mostly" working (it answers calls, dials out, etc. just doesn't detect hangup) my /etc/zaptel.conf is: # # Span Configuration # ~~~~~~~~~~~~~~~~~~ span=1,1,0,esf,b8zs span=2,0,0,esf,b8zs # # Channel Configuration # ~~~~~~~~~~~~~~~~~~~~~ fxsks=1-24 fxoks=25-48 loadzone = us defaultzone=us --CUT-- /etc/asterisk/zapata.conf: [channels] usecallerid=yes callerid=asreceived cidsignalling=bell cidstart=ring callprogress=yes # I have turned this off too ;------------------------------------------------- ; ; Define telco channels in rotary, these should be answered ; like a normal incoming call. ; context=bridgeNEC usecallerid=yes signalling=fxs_ks group=1 ; Part of ZAP group 1 channel => 1-9 context=incoming channel => 12 ;------------------------------------------------- ; ; Telco line, computer dialup, needs to be routed to output line. ; group=2 usecallerid=no channel => 10 ; PSTN attached to Span1:Port10 ;------------------------------------------------- ; ; Telco line, construction trailer fax, needs to be routed. ; group=3 usecallerid=no channel => 11 ; PSTN attached to Span1:Port11 ;------------------------------------------------- ; ; ADTran lines, used for outgoing to analog devices ; context=incoming group=4 usecallerid=no signalling=fxo_ks channel => 25-36 --CUT-- For context, the bridgeNEC context just dials out one of the ADTran lines to our existing NEC system, but the incoming context starts our menu-system, which was also not detecting hangups. I have also tried using loopstart and groundstart signalling, doesn't seem to make a difference. I am pretty well stumped myself. I need to call the telco about the caller id not working to verify that it is still turned on, but I figure I might as well wait so that if I need to ask them about the signalling I can know all the questions to ask at the same time.>Thanks, Daniel
I don't see Magicjack being around long. The business model isn't sustainable without tons of ads, and even then, people will either ignore them if they are audio or if they are popups, they will simply close them or disable them. I might buy one just to hack it. Has anyone sniffed it or poked around at all on lists? Thanks, Steve T On Fri, Jul 11, 2008 at 6:03 PM, C. Savinovich <c.savinovich at itntelecom.com> wrote:> > Yes, I have designed two different webphones, granted, using third party > libraries, and magicjack's quality is better. I acknowledge that. > > Thank you, but referring me to someone's review won't help me much... I am > interested in the internals. Regardless, their technique has a twist, and I > am a naturally very curious *technical* fellow. > > CS > > > -----Original Message----- > From: asterisk-users-bounces at lists.digium.com > [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Joe Greco > Sent: Friday, July 11, 2008 5:41 PM > To: asterisk-users at lists.digium.com > Subject: Re: [asterisk-users] MagicJack quality > >> I am puzzled by the quality of magicjack. I keep trying to figure out how >> they can the quality be that adequate. Since Skype also has an excellent >> quality, that leaves me to believe that software based calls (softphones) >> could have and advantage over hardphones, provided there is a parameter > that >> those 2 companies are addressing. > > You are puzzled by the quality? > > http://www.laptopmag.com/review/voip/magicjack.aspx > > I don't know, but from the sounds of the comments, you'd get about just as > much quality out of an actual cigarette lighter, and probably a good bit > more usefulness. > > Nice EULA, by the way: > > http://gadgets.boingboing.net/2008/04/14/magicjacks-eula-says.html > > VoIP over the Internet isn't /that/ hard. > > ... JG > -- > Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net > "We call it the 'one bite at the apple' rule. Give me one chance [and] then > I > won't contact you again." - Direct Marketing Ass'n position on e-mail > spam(CNN) > With 24 million small businesses in the US alone, that's way too many > apples. > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2008 - September 22 - 25 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2008 - September 22 - 25 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
As Michael Graves points out, people will hack it to run on thin clients and why not virtual machines with very limited access? Maybe an AP with a USB port and OpenWRT or something? Remember when NetZero really cost nothing? I had a program someone wrote to close the as Windows, later I figured out that if I removed a dll file and then just put some junk in a text file and name it whateverthefilewas.dll NetZero was free and I didn't have to use any programs to close the ad windows. Remember the free webhosts that put ads at the bottom of your page but you did get decent free hosting. Remember the scripts that came out within weeks that eliminated those ads? Thanks, Steve T On Fri, Jul 11, 2008 at 6:53 PM, C. Savinovich <c.savinovich at itntelecom.com> wrote:> > As per the ads, if people ignore them or not, doesn't matter. Advertisers > will fall in love with the idea that the venue "reaches" 1 million people, > or more. As per the price of the service, they might be calculating the > fact that the average monthly consumption of minutes on a softphone could be > lower that the average monthly consumption on hardphones. After all, having > to have that cpu on to make the call, is a drag. > > CS > > > -----Original Message----- > From: asterisk-users-bounces at lists.digium.com > [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Steve Totaro > Sent: Friday, July 11, 2008 6:28 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] MagicJack quality > > I don't see Magicjack being around long. The business model isn't > sustainable without tons of ads, and even then, people will either > ignore them if they are audio or if they are popups, they will simply > close them or disable them. > > I might buy one just to hack it. Has anyone sniffed it or poked > around at all on lists? > > Thanks, > Steve T > > > On Fri, Jul 11, 2008 at 6:03 PM, C. Savinovich > <c.savinovich at itntelecom.com> wrote: >> >> Yes, I have designed two different webphones, granted, using third party >> libraries, and magicjack's quality is better. I acknowledge that. >> >> Thank you, but referring me to someone's review won't help me much... I > am >> interested in the internals. Regardless, their technique has a twist, and > I >> am a naturally very curious *technical* fellow. >> >> CS >> >> >> -----Original Message----- >> From: asterisk-users-bounces at lists.digium.com >> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Joe Greco >> Sent: Friday, July 11, 2008 5:41 PM >> To: asterisk-users at lists.digium.com >> Subject: Re: [asterisk-users] MagicJack quality >> >>> I am puzzled by the quality of magicjack. I keep trying to figure out > how >>> they can the quality be that adequate. Since Skype also has an excellent >>> quality, that leaves me to believe that software based calls (softphones) >>> could have and advantage over hardphones, provided there is a parameter >> that >>> those 2 companies are addressing. >> >> You are puzzled by the quality? >> >> http://www.laptopmag.com/review/voip/magicjack.aspx >> >> I don't know, but from the sounds of the comments, you'd get about just as >> much quality out of an actual cigarette lighter, and probably a good bit >> more usefulness. >> >> Nice EULA, by the way: >> >> http://gadgets.boingboing.net/2008/04/14/magicjacks-eula-says.html >> >> VoIP over the Internet isn't /that/ hard. >> >> ... JG >> -- >> Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net >> "We call it the 'one bite at the apple' rule. Give me one chance [and] > then >> I >> won't contact you again." - Direct Marketing Ass'n position on e-mail >> spam(CNN) >> With 24 million small businesses in the US alone, that's way too many >> apples. >> >> _______________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> AstriCon 2008 - September 22 - 25 Phoenix, Arizona >> Register Now: http://www.astricon.net >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> >> >> _______________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> AstriCon 2008 - September 22 - 25 Phoenix, Arizona >> Register Now: http://www.astricon.net >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2008 - September 22 - 25 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2008 - September 22 - 25 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
Hi all, Just in case if anyone will be interested in BroadTel UPA-1, a USB to FXS adapter embedded with SIP softphone. Product specification is as follows: Hardware: USB to RJ11 FXS adapter 1 USB port, for computer connection 1 RJ-11 FXS, for phone connection Dimension (L x W x H): 53 x 15 x 28 mm Support SIP 2.0, IAX2 or Skype built-in 2MB flash drive for storage of BroadTel softphone Support any analog phone or cordless phone Support connecting to PBX Plastic case for the adapter can be designed on a per-client basis Main Features USB to RJ11 FXS converter Support Windows AUTORUN for SELF loading of driver and softphone Embedded SIP, IAX2 softphones or Skype, supporting G.729, G.723.1, G.711(uLaw,aLaw), GSM(FR,AMR), iLBC, Speex and so on Built-in echo cancellation SIP and IAX softphones can be customized with customers'logo Auto detection, installation and configuration of Windows USB audio deveice when UPA-1 is plugged in Support standard windows USB audio device with unique device ID Connect USB phone adapter UPA-1 to your phone, no more microphone and speaker SLIC interface for analog phone connection Support cordless phone set include DECT, 2.4GHz, 900MHz or others Receive SIP, IAX2 or Skype calls by ringing and picking up handset - same as home line Dial SIP, IAX2 and calls through phone pad directly or softphone users interface Connect Skype and SkypeIn calls into PBX or enterprise IVR Dial SkypeOut from PBX digital extension set directly Support Skype speed dial and SkypeOut through phone pad directly Support 1 REN standard loads OEM and ODM orders are welcome. For details, please visit BroadTel corporate web site http://www.broad-tel.com/products/phoneadapter.php , or conact us at broadtel at 126.com if you are interested in the product. Best regards, BroadTel On Sat, Jul 12, 2008 at 5:13 AM, C. Savinovich <c.savinovich at itntelecom.com>wrote:> > I am puzzled by the quality of magicjack. I keep trying to figure out how > they can the quality be that adequate. Since Skype also has an excellent > quality, that leaves me to believe that software based calls (softphones) > could have and advantage over hardphones, provided there is a parameter > that > those 2 companies are addressing. > > Anyone's thoughts on this? > > CS > > > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2008 - September 22 - 25 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110524/2e3951ef/attachment.htm>
On Tuesday 24 May 2011, BroadTel wrote:> Hi all, > Just in case if anyone will be interested in *REDACTED*, a USB to FXS > adapter embedded with SIP softphone. Product specification is as follows:Please refer to the fifth and sixth words of the title of this mailing list. To everyone else, I recommend $ echo -e ":0\n* ^From.*BroadTel\n>/dev/null" >> .procmailrc -- AJS Answers come *after* questions.
On 05/24/2011 11:35 AM, A J Stiles wrote: Someone asked about the quality of it, he was quoting the hardware specs of a similar device. I doubt magicjack publishes that kind of detail about theirs. so where is the problem ? Its irrelevant he represents that device commercially.> On Tuesday 24 May 2011, BroadTel wrote: >> Hi all, >> Just in case if anyone will be interested in *REDACTED*, a USB to FXS >> adapter embedded with SIP softphone. Product specification is as follows: > Please refer to the fifth and sixth words of the title of this mailing list. > > To everyone else, I recommend > $ echo -e ":0\n* ^From.*BroadTel\n>/dev/null">> .procmailrc >