Re: [Asterisk-Users] ilbc and asterisk 1.0.3 - strange noises.
Title: OpSign I am using RedHat 7.2 and this noises on the codec started after I updated GCC to 3.0.4, downgrading it to gcc 2.96 made it work well again. I know, it's time to upgrade de distrubution, but it's running very stable so far, so why change... Thanks. Alessandro Ren wrote: Have someone experienced any strange noises using the ilbc codec after upgrading to asterisk 1.0.3? I had to change the codec do gsm to fix this problem. The noise is very loud, like saturation of the echo ro something, seems like the echo cancelation is amplifying itself. I'be been using ilbs since asterisl 0.70 and have never had any problem like this. Thanks. -- __ AlessandroRen OpServices LucianadeAbreu,471-Sala403 PortoAlegre,RS-CEP90570-060 (phone55(51)3061-3588 4fax55(51)3061-3588 Qmobile55(51)9807-3255 :email[EMAIL PROTECTED] __ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- OpSign __ AlessandroRen OpServices LucianadeAbreu,471-Sala403 PortoAlegre,RS-CEP90570-060 (phone55(51)3061-3588 4fax55(51)3061-3588 Qmobile55(51)9807-3255 :email[EMAIL PROTECTED] __ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ilbc and asterisk 1.0.3 - strange noises.
Title: OpSign Have someone experienced any strange noises using the ilbc codec after upgrading to asterisk 1.0.3? I had to change the codec do gsm to fix this problem. The noise is very loud, like saturation of the echo ro something, seems like the echo cancelation is amplifying itself. I'be been using ilbs since asterisl 0.70 and have never had any problem like this. Thanks. -- __ AlessandroRen OpServices LucianadeAbreu,471-Sala403 PortoAlegre,RS-CEP90570-060 (phone55(51)3061-3588 4fax55(51)3061-3588 Qmobile55(51)9807-3255 :email[EMAIL PROTECTED] __ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ANALOG FXO ZAPTEL WCFXO WCTDM module issues seen with intermittent analog lines
Title: OpSign Do we have ohter alternatiives beseides digium cards? I am having some problems with them too. Thanks Wilson Pickett wrote: cable from one side of the desk to another, and I simply disconnected the RJ-45 connector and plugged it back in. THIS PROMPTLY RESULTED IN VERY VERY SCRATCHY AUDIO CONNECTIONS WHEN USING THE FXO PORT. Incoming calls I had this kind of problem early on too. At the time I rebooted to fix it, but I later observed the driver reload would fix it too. The next step is to imaging that the drivers don't linkstuff being unplugged and replugged when they are running Sorta like changing horses in the middle of the stream :) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- __ AlessandroRen OpServices LucianadeAbreu,471-Sala403 PortoAlegre,RS-CEP90570-060 (phone55(51)3061-3588 4fax55(51)3061-3588 Qmobile55(51)9807-3255 :email[EMAIL PROTECTED] __ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] zaptel and low ring voltage
Title: OpSign I'd plug four telephones in these lines and test if the lines are really engaged or not and in case it is busy, the other will ring or it will bring you to the voicemail. I ha a similiar problem, the telco had no engaged the lines properly, after this was solved , I also had a damaged FXO channel. Can't you replace the card and see what happens? The telco could also have sold more lines that the switch really supports, thus causing sometimes this problem. I have seen this happening with my local telcos. []s. Jim Van Meggelen wrote: [EMAIL PROTECTED] wrote: Hi all, Several months ago we built an * box with a quad-FXO tdm400p (REV e/f). From the get-go, there has been a problem where occasionally (2-3 times a week) zaptel/* will not detect the ringing on a line. (The call will ring through to telco voicemail). The problem is not specific to a single line or FXO port on the tdm400p. I have 2 theories: #1 - the ring voltage for some calls is below acceptable levels Possible, but also possible that there is too much loss on the circuit. You can test the ringing voltage with a meter, it needs to be between 90V and 110V. Beyond that you may need to use a transmission test set (such as a Wilcom T136B). I got mine for $20 bucks on eBay. Using a butt set and the test set you'll need to call a 1004Hz source from TELUS and then check that you're within the following specs: Loop mA: 23 or better (too hot is no good either, but I doubt that's your problem) Circuit loss: between 0 and -8dB. 0 is really too hot, -3 to -6 is nominal, -8.5 is pushing it, but still within spec. #2 - the tdm400p card is bad Assuming #1, can the zaptel driver be tweaked to be more sensitive to ringing? Any other ideas or experiences? Running asterisk/zaptel v1.0.2 Thank you, http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- __ AlessandroRen OpServices LucianadeAbreu,471-Sala403 PortoAlegre,RS-CEP90570-060 (phone55(51)3061-3588 4fax55(51)3061-3588 Qmobile55(51)9807-3255 :email[EMAIL PROTECTED] __ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Interrupt latency problems
Title: OpSign Have any of you tried to disable ACPI on the kernel? Rich Adamson wrote: On Wed, 2004-12-01 at 13:03 -0700, Michael Welter wrote: Steven Critchfield wrote: On Wed, 2004-12-01 at 13:36 -0600, Rich Adamson wrote: So, isn't the issue he/I are chasing after essentially 'why is cpu consumption jumping 30% (or 100%) every ten seconds when zaptel is running with no calls present? So where is that CPU time going? Is it in the system, or userspace? Have you tried changing to a non FC or RH kernel as suggested earlier? Yes, I've just completed the installation of 2.6.9, and the spikes have gone away. Thank you, Steven. Your welcome. I am glad it solved the problem. Now if only someone knew what it was about the stock RH or FC kernel that makes it happen you could get RH or FC to stop using that patch. That or maybe more people will be like me and always cast a weary eye upon a prepackaged kernel no matter what distro it came from. Looking at the Changlog for 2.6.9, it would appear a fair amount of work has been down in the pci stuff and the interrupt support areas. Since that seems to be an issue that keeps rearing its head with the digium analog cards, maybe there is something 'fixed' in that area. Not being a strong linux admin, how difficult would you say installing 2.6.9 is on top of a RHv9 system (2.4.20-31.9) should be for me? Any suggestions/hints on how to do it would be appreciated. Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- __ AlessandroRen OpServices LucianadeAbreu,471-Sala403 PortoAlegre,RS-CEP90570-060 (phone55(51)3061-3588 4fax55(51)3061-3588 Qmobile55(51)9807-3255 :email[EMAIL PROTECTED] __ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fax pass-throught.
Title: OpSign Steve, how would I make transparent for the user ti send a fax via a voip channel? I could not figure this out on your site. Thansk Steve. Steve Prior wrote: Alessandro Ren wrote: I've found the fax extention setting, but this is not what I want to do. I'd like to dial from the line on the other side of the IAX channel to a fax, to cut long distance costs, and send a FAX from the source IAX channel. Like bellow: source destination FAX --- Asterisk --- internet --- Asterisk --- external line - PSTN -- FAX I haven't done this, but I've heard that faxing through a voip connection is problematic. Have you considered the possibility (if you control both Asterisk installations in your diagram) that you could fax to a virtual fax on the source Asterisk system which would capture to a file, email or file transfer the image to the other Asterisk box which would then dial out and send to the final destination? This assumes your source and destination actually have to be real fax machines, otherwise you have even more options. Check out: http://scottstuff.net/scott/archives/000152.html and see if it gives you any ideas. Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- __ AlessandroRen OpServices LucianadeAbreu,471-Sala403 PortoAlegre,RS-CEP90570-060 (phone55(51)3061-3588 4fax55(51)3061-3588 Qmobile55(51)9807-3255 :email[EMAIL PROTECTED] __ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fax pass-throught.
I've already search in the mailing list and voip-wiki site but I can not find any examples in how to send a FAX through a IAX channel. I've found the fax extention setting, but this is not what I want to do. I'd like to dial from the line on the other side of the IAX channel to a fax, to cut long distance costs, and send a FAX from the source IAX channel. Like bellow: source destination FAX --- Asterisk --- internet --- Asterisk --- external line - PSTN -- FAX I've also tried setting the codec to G.711 but it has not worked either. Can anyone shed a light on this matter? I am using Asterisk 1.0.2. Thanks. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
