hank smith
2004-Mar-15 08:52 UTC
Asterisk-Users digest, Vol 1 #3101 - 14 msgs Subject: Re: [Asterisk-Users] Asterisk on KNOPPIX, I have it working, somewhat.
let me know if you get the noppix done I would be interested!!!!! ----- Original Message ----- From: <asterisk-users-request@lists.digium.com> To: <asterisk-users@lists.digium.com> Sent: Sunday, March 14, 2004 7:18 PM Subject: Asterisk-Users digest, Vol 1 #3101 - 14 msgs> Send Asterisk-Users mailing list submissions to > asterisk-users@lists.digium.com > > 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 > asterisk-users-request@lists.digium.com > > You can reach the person managing the list at > asterisk-users-admin@lists.digium.com > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of Asterisk-Users digest..." > > > Today's Topics: > > 1. RE: How to send CallerID trough CAPI ? (Jakob Strebel) > 2. RE: ast_rtp_raw_write errors distorting sound on G729 passthrough(Senad Jordanovic)> 3. Re: Asterisk on KNOPPIX, I have it working, somewhat. (GregBoehnlein)> 4. Re: Radius (Greg Boehnlein) > 5. RE: ast_rtp_raw_write errors distorting sound on G729 passthrough(Senad Jordanovic)> 6. RE: VXML_URL and Cisco 7960 Phones? (Low, Adam) > 7. ISDN PRI A and B, cry for help. (Matthew Branton) > 8. Re: Radius (Derek Bruce) > 9. Asterisk NAT Gateway Setup (Kevin) > 10. Re: Cisco SIP license (Matthew Enger) > 11. Re: ISDN PRI A and B, cry for help. (Steve Underwood) > 12. VoYP.Net: voip directory and ENUM registry (Greg Retkowski) > 13. Re: VoYP.Net: voip directory and ENUM registry (Matt Riddell) > 14. EchoCan (Matt Riddell) > > --__--__-- > > Message: 1 > Date: Sun, 14 Mar 2004 17:49:37 +0100 > To: asterisk-users@lists.digium.com > From: Jakob Strebel <mail@teamstrebel.ch> > Subject: RE: [Asterisk-Users] How to send CallerID trough CAPI ? > Reply-To: asterisk-users@lists.digium.com > > Florian, > > Thanks. Now it works. > jakob > > > >The correct dial syntax for CAPI channels is like this: > > > >CAPI/12345678:b${EXTEN} > > > >where: 12345678 is your outgoing MSN (you would choose 0627775171) and > >${EXTEN} is the number to dial. My mistake was I moved the 'b' too when I > >switched the two numbers around. > > > >Please try Dial(CAPI/0627775171:b123456) or Dial(CAPI/627775171:b123456) > > > This are the relevant sections in the config: > > In extensions.conf > > <snip> > > [globals] > ; > ; globals f=FCr ISDN > CLID=3D0627775171 > <snip> > [outst] > exten =3D> _0.,1,SetCIDNum(${CLID}) > exten =3D> _0.,2,Dial(CAPI/0627775171:b${EXTEN}) ; ok CID is sent correct > <snip> > > In capi.conf > msn=3D0627775171 > > > > --__--__-- > > Message: 2 > From: "Senad Jordanovic" <senad@boltblue.com> > To: <asterisk-users@lists.digium.com> > Subject: RE: [Asterisk-Users] ast_rtp_raw_write errors distorting sound onG729 passthrough> Date: Sun, 14 Mar 2004 17:07:26 -0000 > Reply-To: asterisk-users@lists.digium.com > > Olle E. Johansson wrote: > > Check out the latest CVS, Mark applied changes to the code in this > > area tonight. The rtp.c is changed, so the old patch in > > bugs.digium.com may not be necessary any more. > > > Yes, it is done.. > BUT > Now I get MUCH higher values is the debug messages and can not > understand a word from other party during the conversation. > > Here it is: > Mar 14 17:04:46 DEBUG[262161]: rtp.c:980 ast_rtp_raw_write: Difference > is 1386825128, ms is -1247094945 > Mar 14 17:04:46 DEBUG[262161]: rtp.c:980 ast_rtp_raw_write: Difference > is 1386825608, ms is -1247095005 > Mar 14 17:04:46 DEBUG[262161]: rtp.c:980 ast_rtp_raw_write: Difference > is 1386826088, ms is -1247095065 > Mar 14 17:04:46 DEBUG[262161]: rtp.c:980 ast_rtp_raw_write: Difference > is 1386826560, ms is -1247095124 > > > --__--__-- > > Message: 3 > Date: Sun, 14 Mar 2004 12:51:37 -0500 (EST) > From: Greg Boehnlein <damin@nacs.net> > To: asterisk-users@lists.digium.com > Cc: Jr.richardson@cox.net > Subject: Re: [Asterisk-Users] Asterisk on KNOPPIX, I have it working,somewhat.> Reply-To: asterisk-users@lists.digium.com > > I've got Asterisk running on a minimal install of Debian on a P133 w/ 16 > megs of ram. I can help with the Damn Small Linux side of things, and > perhaps get you out of Dependency hell. > > Do you want the system to be self hosting? I.E. the distribution where > Asterisk lives contains the appropriate compilers, source and includes to > build the system? > > Or do you want a "Development" distribution and a "Target" platform? I.E. > you build on the developmen distro and then run a few scripts to generate > a target ISO w/ the binaries? > > In either case, I've got plenty of resources to offer. I've been toying > with the idea of creating a "Knapterisk" installation for quite some time. > > -- > Vice President of N2Net, a New Age Consulting Service, Inc. Company > http://www.n2net.net Where everything clicks into place! > KP-216-121-ST > > > > > --__--__-- > > Message: 4 > Date: Sun, 14 Mar 2004 13:01:26 -0500 (EST) > From: Greg Boehnlein <damin@nacs.net> > To: asterisk-users@lists.digium.com > Subject: Re: [Asterisk-Users] Radius > Reply-To: asterisk-users@lists.digium.com > > On Sat, 13 Mar 2004, Derek Bruce wrote: > > > I have just uploaded a new res_radius package to the bugtracker at > > http://bugs.digium.com/bug_view_page.php?bug_id=0001193 > > > > It has configuration examples this time... and no longer requires > > installation of freeradius. > > Derek, > You package is interesting and took me a while to get a handle on, > but I understand what you are attempting to do with it. I've been > considering the Radius Accounting side of the picture for a while now, and > I see that you have some logic in your system to handle some very basic > attributes for call accounting. Have you considered expanding them and > splitting out the CDR side of the equation into a "cdr_radius" module? It > seems to me that just having a basic cdr to radius gateway for accounting > would be a logical way to accomplish this. Your code appears to have all > of the neccessary elements forming the basis for doing that. > > -- > Vice President of N2Net, a New Age Consulting Service, Inc. Company > http://www.n2net.net Where everything clicks into place! > KP-216-121-ST > > > > > --__--__-- > > Message: 5 > From: "Senad Jordanovic" <senad@boltblue.com> > To: <asterisk-users@lists.digium.com> > Subject: RE: [Asterisk-Users] ast_rtp_raw_write errors distorting sound onG729 passthrough> Date: Sun, 14 Mar 2004 18:07:39 -0000 > Reply-To: asterisk-users@lists.digium.com > > Michael Shuler wrote: > > Thanks, I know though, I paid for it to be done ;) And the whole > > community gets to benefit from it, guess that's my contribution back > > to the open source world that I use all the time to make money with > > :) > > > For whatever is worth.. THANKS :) > > > > --__--__-- > > Message: 6 > From: "Low, Adam" <ALow@Prioritytelecom.com> > To: "'asterisk-users@lists.digium.com'" <asterisk-users@lists.digium.com> > Subject: RE: [Asterisk-Users] VXML_URL and Cisco 7960 Phones? > Date: Sun, 14 Mar 2004 19:08:25 +0100 > Reply-To: asterisk-users@lists.digium.com > > I tried to get that working as well and also found it was not available inthe SIP image. You can't do pushes either to the phone like you can with SCCP.> > -----Original Message----- > From: Rich Adamson [mailto:radamson@routers.com] > Sent: 14 March 2004 13:27 > To: asterisk-users@lists.digium.com > Subject: Re: [Asterisk-Users] VXML_URL and Cisco 7960 Phones? > > > > > I was tempted by the wiki that mentions the (very undocumented) VXML_URL > > and suggests it might be able to control the display on a Cisco phone > > during an incoming call using a SIP image. > > > > I've mucked around with this for over two hours and after scouringsource> > code, google, and the archives have found nothing. > > > > Does anyone have any how to use this feature? Does it even really exist?I> > can see the header being set and hitting the phone - but I can't find > > documentation anywhere suggesting what format you can send it. > > It's my understanding, although I've no direct experience, the function > does not exist in the SIP images. > > The limitation is highly likely related to Cisco marketing plans and not > to real design/programming capability, etc. (How else would one sell > proprietary systems?) > > Anyone have a disassembler? > > > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > ********* DISCLAIMER ********* > > This message and any attachment are confidential and may be privileged orotherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person> > > > --__--__-- > > Message: 7 > From: Matthew Branton <mbranton@xtracard.com> > To: "Asterisk-Users (E-mail)" <asterisk-users@lists.digium.com> > Date: Sun, 14 Mar 2004 13:43:24 -0500 > Subject: [Asterisk-Users] ISDN PRI A and B, cry for help. > Reply-To: asterisk-users@lists.digium.com > > This message is in MIME format. Since your mail reader does not understand > this format, some or all of this message may not be legible. > > ------_=_NextPart_001_01C409F4.3B5E6A10 > Content-Type: text/plain; > charset="iso-8859-1" > > Does asterisk support PRI protocol version A? > Right now I have it working great with B, but in adding some new telcolines> I realize that they are spitting out protocol version A, any way to get > asterisk to start talking A? > > Any help would be really appreciated! > > > Matt > > ------_=_NextPart_001_01C409F4.3B5E6A10 > Content-Type: text/html; > charset="iso-8859-1" > Content-Transfer-Encoding: quoted-printable > > <!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 3.2//EN"> > <HTML> > <HEAD> > <META HTTP-EQUIV=3D"Content-Type" CONTENT=3D"text/html; > charset=3Diso-8859-1"> > <META NAME=3D"Generator" CONTENT=3D"MS Exchange Server version > 5.5.2653.12"> > <TITLE>ISDN PRI A and B, cry for help.</TITLE> > </HEAD> > <BODY> > > <P><FONT SIZE=3D2 FACE=3D"Arial">Does asterisk support PRI protocol > version A?</FONT> > <BR><FONT SIZE=3D2 FACE=3D"Arial">Right now I have it working great > with B, but in adding some new telco lines I realize that they are > spitting out protocol version A, any way to get asterisk to start > talking A?</FONT></P> > > <P><FONT SIZE=3D2 FACE=3D"Arial">Any help would be really > appreciated!</FONT> > </P> > <BR> > > <P><FONT SIZE=3D2 FACE=3D"Arial">Matt</FONT> > </P> > > </BODY> > </HTML> > ------_=_NextPart_001_01C409F4.3B5E6A10-- > > --__--__-- > > Message: 8 > From: "Derek Bruce" <dbruce@calgarytelecom.com> > To: <asterisk-users@lists.digium.com> > Subject: Re: [Asterisk-Users] Radius > Date: Sun, 14 Mar 2004 16:22:39 -0700 > Organization: Calgary Telecom > Reply-To: asterisk-users@lists.digium.com > > Yes, I did consider a cdr_radius approach... In fact, that was my original > attempt... > > That approach quickly proved to be problematic for a few reasons: > 1) Our radius server does 'live' updates to our client database. ie: callis> placed, customer can see the call in progress live. call is completed, the > radius server updates the customers account balance. This proved to be a > problem since asterisk processed calls that it shouldn't have (ie: call is > transfered between multiple call legs or protocols) and this causedtrouble> with multiple radius entries per call. This is aleviated by having thedial> app 'talk' to the cdr app with the cdr userfield, allowing the cdr app the > ability to process cdrs for only the originating call leg. > 2) Radius accounting is a 2 (or 3) part process... START and STOP records > (and possibly ALIVE messages). Having a CDR only solution prevents 'live' > monitoring of call procession... something my users have become accustomed > to. > 3) For my specific application, there was a need to be compatible with our > existing prepid/postpaid calling platform, and having asterisk do it inthe> same manner was desirable. > > Having the cdr application only does work... but entails more 'back end' > work from a billing perspective... such as finding and consolidating cdrfor> multiple call legs, adjusting account balances (if your radius server does > automatic account updates. > > I'm currently working on a more robust and generic method of handling the > mapping of radius responses to internally used variables... > > > ----- Original Message ----- > From: "Greg Boehnlein" <damin@nacs.net> > To: <asterisk-users@lists.digium.com> > Sent: Sunday, March 14, 2004 11:01 AM > Subject: Re: [Asterisk-Users] Radius > > > > Derek, > > You package is interesting and took me a while to get a handle on, > > but I understand what you are attempting to do with it. I've been > > considering the Radius Accounting side of the picture for a while now,and> > I see that you have some logic in your system to handle some very basic > > attributes for call accounting. Have you considered expanding them and > > splitting out the CDR side of the equation into a "cdr_radius" module?It> > seems to me that just having a basic cdr to radius gateway foraccounting> > would be a logical way to accomplish this. Your code appears to have all > > of the neccessary elements forming the basis for doing that. > > > > -- > > Vice President of N2Net, a New Age Consulting Service, Inc. Company > > http://www.n2net.net Where everything clicks into place! > > KP-216-121-ST > > > > > > > > _______________________________________________ > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > --__--__-- > > Message: 9 > From: "Kevin" <Asterisk@gtcus.com> > To: <asterisk-users@lists.digium.com> > Date: Sun, 14 Mar 2004 18:28:11 -0500 > Subject: [Asterisk-Users] Asterisk NAT Gateway Setup > Reply-To: asterisk-users@lists.digium.com > > I am currently using Asterisk behind Belkin NAT router. With what ever > NAT router I have used, I have had difficulties in registration and > audio problems with my SIP provider (Iconnect and Nikotel) > > It was suggested that I try to connect the asterisk box directly to the > internet avoiding the NAT transition. As I will still need internet > connectivity, I am trying to make the asterisk box the NAT gateway. > > I have an additional NIC for my Asterisk box. > > As I am no Linux or Asterisk expert, can anyone make suggestions as to > this approach and any recommended steps to accomplish this? > > Also, how would Asterisk know which interface to bind to? I know there > is a bindaddress= parameter in the SIP config, but the address to the > internet is dynamic via DHCP from my cable provider. > > Thanks, > > Kevin > > > > > > --__--__-- > > Message: 10 > Subject: Re: [Asterisk-Users] Cisco SIP license > From: Matthew Enger <menger@xi.com.au> > To: asterisk-users@lists.digium.com > Organization: Xintegration > Date: 15 Mar 2004 10:34:13 +1100 > Reply-To: asterisk-users@lists.digium.com > > Hello, > > I had the same problem, the license is bound to the phone, apart from > the record on the invoice, I had no proof of the product being > purchased. > > By default cisco ships the phone with skinny images. To get SIP images > without cco access, file a TAC case and they will give you the images. > That is what I had to do. > > Hope this helps, > > Matthew Enger > m.enger@xi.com.au > > On Sat, 2004-03-13 at 10:13, Michael Welter wrote: > > A few days ago the 7960 phones were delivered. Today the received the > > power adapters. However, we've seen nothing about the SIP licenses > > (which were bundled into the price.) > > > > Does anyone have a tftp site that I can use to download the firmware. I > > would like to use this site until Lewan Assoc. sorts the license issue. > > > > Thanks, > > Mike > > > > > > _______________________________________________ > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > -- > Matthew Enger <menger@xi.com.au> > Xintegration > > > --__--__-- > > Message: 11 > Date: Mon, 15 Mar 2004 08:31:37 +0800 > From: Steve Underwood <steveu@coppice.org> > To: asterisk-users@lists.digium.com > Subject: Re: [Asterisk-Users] ISDN PRI A and B, cry for help. > Reply-To: asterisk-users@lists.digium.com > > Hi Matthew, > > I've encountered lots of ISDN protocols, many of which are now obsolete. > I haven't heard of ones called PRI-A or PRI-B. Is this some local > terminology? There is plenty of that. > > Regards, > Steve > > > Matthew Branton wrote: > > > Does asterisk support PRI protocol version A? > > Right now I have it working great with B, but in adding some new telco > > lines I realize that they are spitting out protocol version A, any way > > to get asterisk to start talking A? > > > > Any help would be really appreciated! > > > > > > Matt > > > > > --__--__-- > > Message: 12 > Date: Sun, 14 Mar 2004 16:44:58 -0800 (PST) > From: Greg Retkowski <greg@rage.net> > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] VoYP.Net: voip directory and ENUM registry > Reply-To: asterisk-users@lists.digium.com > > A quick announcement for a project I have started; VoYP.Net, a VoIP > yellow-pages directory & ENUM service. Take a look, register your voip > contact info. I am seeking feedback on it's usability and utility! > > This is the initial public test release of VoYP.Net. The site provides > directory for voip-accessable resources on the Internet. It is set up in a > hierarchy which can be browsed or searched. Users can search the database > by their area code limiting results to geographically local resources. > > VoYP.Net also implements ENUM. ENUM is an IETF-released protocol which > allows analog telephone numbers to be translated to VoIP resources. It > allows calls using analog telephone numbers to be routed entirely over the > internet avoiding telephone charges. > > -- Greg > > Greg Retkowski / I.T. Infrastructure Consultant /)/|//` > greg@rage.net http://www.rage.net/~greg/ C:408-455-3913 /|/ /_/ > > > --__--__-- > > Message: 13 > From: "Matt Riddell" <matt@surecall.co.nz> > To: <asterisk-users@lists.digium.com> > Subject: Re: [Asterisk-Users] VoYP.Net: voip directory and ENUM registry > Date: Mon, 15 Mar 2004 14:23:43 +1300 > Reply-To: asterisk-users@lists.digium.com > > Only problem I can see if that currently, I can search for a or e etc...if > this gives me a list of urls/numbers, what is to stop me feeding them intoa> database to telemarket to them? > > As far as I'm aware there is no internation Do Not Call registry for > internet based calls... > > Anyone know anymore about this? > > Anyone keen to help with this? > > Would it be worthwhile? > > Is there another? > > Kind regards, > > Matt > > P.S. I don't really want to do it but someone should and if noone elsewants> to, I will... > > ----- Original Message ----- > From: "Greg Retkowski" <greg@rage.net> > To: <asterisk-users@lists.digium.com> > Sent: Monday, March 15, 2004 1:44 PM > Subject: [Asterisk-Users] VoYP.Net: voip directory and ENUM registry > > > > A quick announcement for a project I have started; VoYP.Net, a VoIP > > yellow-pages directory & ENUM service. Take a look, register your voip > > contact info. I am seeking feedback on it's usability and utility! > > > > This is the initial public test release of VoYP.Net. The site provides > > directory for voip-accessable resources on the Internet. It is set up ina> > hierarchy which can be browsed or searched. Users can search thedatabase> > by their area code limiting results to geographically local resources. > > > > VoYP.Net also implements ENUM. ENUM is an IETF-released protocol which > > allows analog telephone numbers to be translated to VoIP resources. It > > allows calls using analog telephone numbers to be routed entirely overthe> > internet avoiding telephone charges. > > > > -- Greg > > > > Greg Retkowski / I.T. Infrastructure Consultant /)/|//` > > greg@rage.net http://www.rage.net/~greg/ C:408-455-3913 /|/ /_/ > > > > _______________________________________________ > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > --__--__-- > > Message: 14 > From: "Matt Riddell" <matt@surecall.co.nz> > To: <asterisk-users@lists.digium.com> > Date: Mon, 15 Mar 2004 15:04:40 +1300 > Subject: [Asterisk-Users] EchoCan > Reply-To: asterisk-users@lists.digium.com > > Anyone know why Mark2 is default not Mark3? > > What is the difference? > > I dial work (I'm using DIAX on win98, them Zap chans) I hear no echo, but > they can hear their voice.... > > This is not a problem on other calls they make... > > I have echocancel set to yes and training too > > Any ideas? > > Matt > > > > --__--__-- > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > > > End of Asterisk-Users Digest