Does Asterisk support Radius accounting?.... -----Mensaje original----- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de [EMAIL PROTECTED] Enviado el: Lunes, 17 de Noviembre de 2003 12:08 p.m. Para: [EMAIL PROTECTED] Asunto: Asterisk-Users digest, Vol 1 #1912 - 11 msgs
Send Asterisk-Users mailing list submissions to [EMAIL PROTECTED] To subscribe or unsubscribe via the World Wide Web, visit http://lists.digium.com/mailman/listinfo/asterisk-users or, via email, send a message with subject or body 'help' to [EMAIL PROTECTED] You can reach the person managing the list at [EMAIL PROTECTED] When replying, please edit your Subject line so it is more specific than "Re: Contents of Asterisk-Users digest..." Today's Topics: 1. RE: FXO Cards in Australia (Gonzalo Servat) 2. Re: IAX2 connectivity problem (qualify=yes) (Philipp von Klitzing) 3. RE: FXO Cards in Australia (Bryan Nolen) 4. RE: FXO Cards in Australia (Gonzalo Servat) 5. Re: Meetme : Zaptel ztdummy errors (Andrew Thompson) 6. SIP soft phone registration (Steve Murphy) 7. Re: DTMF (Sean P. Robertson) 8. VOIP phonesets vs. cheap Analog touch-tone sets with Asterisk (Steve Murphy) 9. Re: NuFone International Calls (marrandy) 10. Re: Meetme : Zaptel ztdummy errors ([EMAIL PROTECTED]) 11. iconnecthere incoming ([EMAIL PROTECTED]) --__--__-- Message: 1 Subject: RE: [Asterisk-Users] FXO Cards in Australia From: Gonzalo Servat <[EMAIL PROTECTED]> To: [EMAIL PROTECTED] Organization: Webtastic Date: Tue, 18 Nov 2003 00:17:30 +1100 Reply-To: [EMAIL PROTECTED] On Mon, 2003-11-17 at 23:53, Adam Goryachev wrote: > Yes, echo problems do still exist, I would suggest testing it before > going live. Yeah, so I've heard. > A couple of points to note: > 1) Using soft phones seems to compound the issue So the echo problems are not so bad when using software phones? > 2) A faster CPU seems to help (I upgraded from a PII300 to a Athlon > 2200) > 3) When dialling in/out over the ISDN DTMF won't work (at least I > haven't seen the patch which purportedly allows it to work) when you > use the isdn4linux patch. This is specific to the NetJet card once again, right? Time to go hunting for the patch... > 4) Without the above kernel patch you will hear DTMF tones instead of > the other persons voice when they talk. They don't hear the tones or > notice anything wrong. Hmm, not good. Since we want to run a small IVR the DTMF tones are kinda needed. > In short, if you can live with the above problems, then you can get > away with it, from what I know now, I would suggest getting a > chan_capi capable device, though I haven't tried that yet. The NetJet is supposedly CAPI capable. Have you tried installing this? --> http://www.junghanns.net/asterisk/page1.html > I am about to switch from a netjet card to a TE4xxP card as soon as > possible, I have a OnRamp 10 being installed tomorrow. This is largely > to increase the number of incoming lines, but partly to resolve the > above issues, and also partly to try to resolve long running > reliability issues which may in fact be related to the TDM400P anyway. > In which case I will be looking for a T1 channel bank some time soon > :( Argh, the fun never stops :) > PS, I have a brand new Traverse Netjet card available (it was to be > used for a dial-up ISDN internet account) which is no longer needed. How much do you want for it? If you can confirm whether the capi channel driver works with it and reduces the echo problem, I'll be interested. Thanks for your help. Regards, Gonzalo --__--__-- Message: 2 Date: Mon, 17 Nov 2003 14:28:14 +0100 From: Philipp von Klitzing <[EMAIL PROTECTED]> Subject: Re: [Asterisk-Users] IAX2 connectivity problem (qualify=yes) To: [EMAIL PROTECTED] Organization: AEGEE Reply-To: [EMAIL PROTECTED] Hi! > Try "qualify=3000" which will increase the time between checks.. > Although it sounds like there is more to this problem than just > increasing the time.. That's not really what I want to do - quality is really bad if you go above 2000, so it makes sense to keep it at this. I can schedule a reload, of course, e.g. once an hour, but that can't be the correct way to do this... is there really no-one else out here who has seen this?? Cheers, Philipp --__--__-- Message: 3 From: "Bryan Nolen" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Subject: RE: [Asterisk-Users] FXO Cards in Australia Date: Tue, 18 Nov 2003 00:48:34 +1100 Reply-To: [EMAIL PROTECTED] Re: these problems with the NetJet Cards: have people spoken with Traverse about them? I have found them to be most helpful with any problems (mainly with the Pulsar PCI ADSL cards) Try talking to [EMAIL PROTECTED] ? -Bryan > -----Original Message----- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > Gonzalo Servat > Sent: Tuesday, 18 November 2003 12:18 AM > To: [EMAIL PROTECTED] > Subject: RE: [Asterisk-Users] FXO Cards in Australia > > > On Mon, 2003-11-17 at 23:53, Adam Goryachev wrote: > > Yes, echo problems do still exist, I would suggest testing it before > > going live. > > Yeah, so I've heard. > > > A couple of points to note: > > 1) Using soft phones seems to compound the issue > > So the echo problems are not so bad when using software phones? > > > 2) A faster CPU seems to help (I upgraded from a PII300 to a Athlon > > 2200) > > 3) When dialling in/out over the ISDN DTMF won't work (at least I > > haven't seen the patch which purportedly allows it to work) > when you use > > the isdn4linux patch. > > This is specific to the NetJet card once again, right? Time to go > hunting for the patch... > > > 4) Without the above kernel patch you will hear DTMF tones > instead of > > the other persons voice when they talk. They don't hear the tones or > > notice anything wrong. > > Hmm, not good. Since we want to run a small IVR the DTMF > tones are kinda > needed. > > > In short, if you can live with the above problems, then you > can get away > > with it, from what I know now, I would suggest getting a chan_capi > > capable device, though I haven't tried that yet. > > The NetJet is supposedly CAPI capable. Have you tried installing this? > --> http://www.junghanns.net/asterisk/page1.html > > > I am about to switch from a netjet card to a TE4xxP card as soon as > > possible, I have a OnRamp 10 being installed tomorrow. This > is largely > > to increase the number of incoming lines, but partly to resolve the > > above issues, and also partly to try to resolve long > running reliability > > issues which may in fact be related to the TDM400P anyway. > In which case > > I will be looking for a T1 channel bank some time soon :( > > Argh, the fun never stops :) > > > PS, I have a brand new Traverse Netjet card available (it > was to be used > > for a dial-up ISDN internet account) which is no longer needed. > > How much do you want for it? If you can confirm whether the > capi channel > driver works with it and reduces the echo problem, I'll be interested. > > Thanks for your help. > > Regards, > Gonzalo > > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > --__--__-- Message: 4 Subject: RE: [Asterisk-Users] FXO Cards in Australia From: Gonzalo Servat <[EMAIL PROTECTED]> To: [EMAIL PROTECTED] Organization: Webtastic Date: Tue, 18 Nov 2003 01:10:06 +1100 Reply-To: [EMAIL PROTECTED] I'll be speaking to Guy tomorrow about this. Guy is certainly a helpful & friendly guy and I'm sure he'll be keen to hear about these echo problems. Regards, Gonzalo On Tue, 2003-11-18 at 00:48, Bryan Nolen wrote: > Re: these problems with the NetJet Cards: have people spoken with > Traverse about them? I have found them to be most helpful with any > problems (mainly with the Pulsar PCI ADSL cards) > > Try talking to [EMAIL PROTECTED] ? > > -Bryan > > > -----Original Message----- > > From: [EMAIL PROTECTED] > > [mailto:[EMAIL PROTECTED] On Behalf Of > > Gonzalo Servat > > Sent: Tuesday, 18 November 2003 12:18 AM > > To: [EMAIL PROTECTED] > > Subject: RE: [Asterisk-Users] FXO Cards in Australia > > > > > > On Mon, 2003-11-17 at 23:53, Adam Goryachev wrote: > > > Yes, echo problems do still exist, I would suggest testing it > > > before going live. > > > > Yeah, so I've heard. > > > > > A couple of points to note: > > > 1) Using soft phones seems to compound the issue > > > > So the echo problems are not so bad when using software phones? > > > > > 2) A faster CPU seems to help (I upgraded from a PII300 to a > > > Athlon > > > 2200) > > > 3) When dialling in/out over the ISDN DTMF won't work (at least I > > > haven't seen the patch which purportedly allows it to work) > > when you use > > > the isdn4linux patch. > > > > This is specific to the NetJet card once again, right? Time to go > > hunting for the patch... > > > > > 4) Without the above kernel patch you will hear DTMF tones > > instead of > > > the other persons voice when they talk. They don't hear the tones > > > or notice anything wrong. > > > > Hmm, not good. Since we want to run a small IVR the DTMF > > tones are kinda > > needed. > > > > > In short, if you can live with the above problems, then you > > can get away > > > with it, from what I know now, I would suggest getting a chan_capi > > > capable device, though I haven't tried that yet. > > > > The NetJet is supposedly CAPI capable. Have you tried installing > > this? > > --> http://www.junghanns.net/asterisk/page1.html > > > > > I am about to switch from a netjet card to a TE4xxP card as soon > > > as possible, I have a OnRamp 10 being installed tomorrow. This > > is largely > > > to increase the number of incoming lines, but partly to resolve > > > the above issues, and also partly to try to resolve long > > running reliability > > > issues which may in fact be related to the TDM400P anyway. > > In which case > > > I will be looking for a T1 channel bank some time soon :( > > > > Argh, the fun never stops :) > > > > > PS, I have a brand new Traverse Netjet card available (it > > was to be used > > > for a dial-up ISDN internet account) which is no longer needed. > > > > How much do you want for it? If you can confirm whether the > > capi channel > > driver works with it and reduces the echo problem, I'll be interested. > > > > Thanks for your help. > > > > Regards, > > Gonzalo > > > > _______________________________________________ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users -- ========================================= WEBTASTIC ABN 30 087 960 619 PO Box 3024 Willoughby North, NSW, 2068 Phone: +61 (02) 9499 2452 Fax: +61 (02) 9499 2618 Email: [EMAIL PROTECTED] Web: http://www.webtastic.com.au ========================================= --__--__-- Message: 5 From: "Andrew Thompson" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Subject: Re: [Asterisk-Users] Meetme : Zaptel ztdummy errors Date: Mon, 17 Nov 2003 09:09:30 -0500 Reply-To: [EMAIL PROTECTED] 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Company Date: 17 Nov 2003 07:14:25 -0700 Subject: [Asterisk-Users] SIP soft phone registration Reply-To: [EMAIL PROTECTED] Hello-- I've installed a few X-Lite softphones on windows machines, and am playing with the settings. I've written before about this, and since have discovered that the X-Lite has an option to turn off registration, which I have set, because I bumped into a letter from Mark, saying that Asterisk doesn't do registrations yet. And, without such on the X-Lite, I can still dial out, and get calls. And, best of all, no 60-second repetition of the same error message over and over and over to fill up the logs and the screens. My question: What is this feature? What is SIP registration? murf --__--__-- Message: 7 From: "Sean P. Robertson" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Subject: Re: [Asterisk-Users] DTMF Date: Mon, 17 Nov 2003 09:24:45 -0500 Reply-To: [EMAIL PROTECTED] I think that you are thinking of SIP INFO messages if you are expecting to see something in the SIP messaging. RFC2833 is sent as part of the RTP packets so you are not going to see a plain text 1,2,3,4,etc in a trace when using it. http://www.faqs.org/rfcs/rfc2833.html Sean ----- Original Message ----- From: "Scott England" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Monday, November 17, 2003 5:58 AM Subject: [Asterisk-Users] DTMF > I am trying to connect to a vocal server from an asterisk server. A > call is received via iax2 to my asterisk server. I then initiate a SIP > connection to the vocal server. everything works great except dtmf > doesnt work. A cisco 5300 can connect to this vocal server and do dtmf > without a problem. I have my dtmf set to rfc2833 in the general > section of the sip.conf . I can confirm that the channel is in rfc2833 > during the call via show channel. With SIP debug though I dont see any > event for dtmf. I do see dtmf in IAX though. > > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > --__--__-- Message: 8 From: Steve Murphy <[EMAIL PROTECTED]> To: [EMAIL PROTECTED] Organization: Electronic Tools Company Date: 17 Nov 2003 07:34:02 -0700 Subject: [Asterisk-Users] VOIP phonesets vs. cheap Analog touch-tone sets with Asterisk Reply-To: [EMAIL PROTECTED] Hello-- I've been asked an interesting question, and I'm too ignorant to answer it authoritatively (yet). Can anyone help me? Question: If I'm going to implement a somewhat small (10-80) phone system, and I have a choice of using VOIP phoneset (like SNOM or Grandstream or Cisco, etc), vs. cheap analog touch-tone phones, exactly what features will I kiss goodbye if I use the cheap analogs? In other words, what features will a (more expensive) VOIP phoneset provide, that the analog won't? I know already that asterisk will give me these features with just plain analog phones (&zaptel cards, of course): Voice mail, park & retrieve & MOH, transfer, agents, and a few others. And, if you get an analog with a CID built in, you could have that, too? (Haven't tried that yet). What is the justification for VOIP? just total cost reductions (if the phone is cheap enough?)? Or are there some nicenesses that only VOIPs can supply? murf --__--__-- Message: 9 From: marrandy <[EMAIL PROTECTED]> To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] NuFone International Calls Date: Mon, 17 Nov 2003 09:43:46 -0500 Reply-To: [EMAIL PROTECTED] On Sunday 16 November 2003 12:52 pm, [EMAIL PROTECTED] wrote: Hello Jasim. > lag just took on a whole new definition :) -- Old mail I stored for the future for when I have time to look at asterisk again. Like now :-) Basically, I'm going over the scripts/files and resetting and testing. This is still a test system, but I thought I would give some feedback. > in iax.conf put > > [nufone] > type=peer > secret=yourpassword > context=WORLD ; <-- this bit originally stated, maynot be necessary > host=switch-1.nufone.net disallow=all > allow=ilbc > trunk=yes > > in extensions.conf put > > exten => _011.,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN}|60) I didn't need the context=WORLD My extensions.conf has the following for international calls. Any comments appreciated :- [globals] TRUNK=IAX2/[EMAIL PROTECTED] [int-trunks] ; exten => _9011.,1,Dial(${TRUNK}/${EXTEN:1}) exten => _9011.,2,Congestion I found I had echo in that I could hear my voice about a third of a second later. It turned out that one of the many sets of scripts I had been testing had a high tx and rx gain setting. When I changed the settings to :- echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=0.0 txgain=0.0 in /etc/asterisk/zapata.conf the issue went away. YMMV > now, jerjer says you don't need CONTEXT in your iax.conf anymore, he > handles it at his end, but this is a config from donkeys ages ago and how > we used to do things in the elden days... > > - wasim > > p.s. for those in the know ... the super secret code name for eeks has > been herewith changed to farfon (with two dots on the "o" of the fon) Ahhh...secrets...What's it got in its pockets ? Regards...Martin -- Even God lends a hand to honest boldness. -- Menander --__--__-- Message: 10 Date: Mon, 17 Nov 2003 09:58:40 -0500 (EST) From: [EMAIL PROTECTED] To: Asterisk-Users Mailing-list <[EMAIL PROTECTED]> Subject: Re: [Asterisk-Users] Meetme : Zaptel ztdummy errors Reply-To: [EMAIL PROTECTED] HI guys I do have usb-uhci. How do I build ztdummy? I think once its built I just have to do a modprobe to load it, I just don't know how to load it. AJ --__--__-- Message: 11 Date: Mon, 17 Nov 2003 10:12:22 -0500 (EST) From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: [Asterisk-Users] iconnecthere incoming Reply-To: [EMAIL PROTECTED] Hi guys I just registered an incoming number with iconnecthere and I'm trying to set up incoming calls from icconnecthere on my asterisk server. I took a look at john todds sample sip.conf and extensions.conf file but for some reason my incoming is still not working. At this point I wish to use iconnecthere merely for inbound calls. Also my asterisk server is behind nat. The following is my current sip.conf file or at least the part that I see as pertinent here. [general] register=15555555555:[EMAIL PROTECTED]/15555555555 context = sip-incoming The pertinent part of my current extensions.conf is this [sip-incoming] exten => _.,1,Goto(home,s,1) Please note that 15555555555 above has been substituted for my 11 digit iconnecthere number and 1111 above has been substituted for my secret/password. I have seen on the mailing list where nat = 1 seems to be used by some to get this working and I even saw this in john's sip.conf file; however it was not in the general section. Can I place this in the general section? Thanks a lot for any and all assistance. AJ --__--__-- _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users End of Asterisk-Users Digest _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users