Re: [Asterisk-Users] ADSI phone vs. IP phone
Aastra will have a production PT480i SIP phone in March for ~US180-$200. Same phone as ADSI model just SIP, but has 4 extra buttons for virtual lines. Got a beta SIP model under test. Designed for SIP v1 v2. * is one of PBX used for testing by development, so should be * friendly when released. For all those curious, and i'd bet more like us$250 :) http://www.sayson.com/product/voip_phone.htm ___ 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] SS7 over Asterisk ?
Yes , you are right All I need is an IP cloud to transport some E1's from an PBX to another... Regards Alex - Original Message - From: Tom Scott [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, January 19, 2004 2:45 PM Subject: Re: [Asterisk-Users] SS7 over Asterisk ? Alexandru, I think the subject line has a tendency to confuse the issue we're discussing here. At least remove SS7 from it and call it, maybe, TDMoIP, TDMoPW (it's actually a pseudo wire you're looking for, i think). You want to transport E1 over an IP cloud, right? You don't want the IP cloud to handle the call routing, only to carry the E1 from one PBX to another. But maybe I misunderstand. In any case, I doubt if you need to deal with SS7. And if you want to do TDMoPW, I doubt that asterisk is designed to handle it, since asterisk handles is call oriented. You already have PBXs to handle the calls; all you need is an IP could (or ATM, or MPLS) to carry E1 pipes. But as I said, maybe i misunderstand. -- TT Alexandru Coseru wrote: Maybe , I never tried TDMoE ... Where can I found a documentation or at least a sample for doing that ? Second , there is a small problem... Their are not on the same subnet, but this can be fixed(i hope) using tunneling.. Regards Alex - Original Message - From: Nicolas Bougues [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, January 19, 2004 10:05 AM Subject: Re: [Asterisk-Users] SS7 over Asterisk ? On Sat, Jan 17, 2004 at 04:34:34PM +0200, Alexandru Coseru wrote: All I'm trying right now is to get raw data from the E1 (from each timeslot) , transmit it to another asterisk server and push it to the other E1.. Doesn't TDMoE do that (provided that you're on the same subnet) ? -- Nicolas Bougues Axialys Interactive ___ 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 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] [ot] Grandstream hardware
On Mon, Jan 19, 2004 at 05:23:26PM -0600, Eric Wieling wrote: How does Grandstream become patent indemnified for their hardware? I would assume they did not pay for a license for G723,1 and G729 directly to the patent holding company. Maybe they did. I always assumed the indemnification came with a DSP that implemented the codec. I suppose they did pay for it. A DSP is a processor. Just like when you buy a Pentium IV, it doesn't give you the right to use, for instance, MS Windows on it. You have to pay for software. And that's what algorithms are. Except that you have to pay for algorithms even if you do your own original implementation. -- Nicolas Bougues Axialys Interactive ___ 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] SS7 over Asterisk ?
Now I understood that I can't use Asterisk for what I'm planning to do.. By using TDMoE , all I get is a new span on the destination server... But from there , I'm stuck... Anyway , thanks Regards Alex - Original Message - From: Philipp von Klitzing [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, January 19, 2004 8:07 PM Subject: Re: [Asterisk-Users] SS7 over Asterisk ? Hi! Maybe , I never tried TDMoE ... Where can I found a documentation or at least a sample for doing that ? http://www.asteriskdocs.org/current/docs-pdf/hgta.pdf page 29 Note that this book is still in pre-alpha state... Philipp ___ 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 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] user password and call waiting
Can you give me an example or point me to the page where account codes are described? Thanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian West Sent: Tuesday, January 20, 2004 1:08 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] user password and call waiting Use account codes. That works ALOT better. If you require passwords then look at app_authenticate. bkw On Tue, 20 Jan 2004, Ing Isianto Istiadi wrote: Dear all, I have a questions: 1. I have 3 FXO connected to 3 analog phones. But I have 5 users using those phone. I want to be able to log who is using the phones and where to. I'd like to use password for each user so that I can keep track who is the caller and for how long. I read the authenticate application, but I think it is for one user only. Forgive my English. Fxo -- phone1 user A use phone1 or phone2 or phone3 after entering Fxo -- phone2 password like 1234, so if A want to call from either phones Fxo -- phone3 A needs to punch 91234xxx The same with user B, B needs to punch 92345xx And so on. But in my logger (either text based or database based), I need to see the caller is A and the rest is the same. Can I do this with *. What is the effective approach? 2. I Use digium hardware (FXO and FXS), * v0.5. Can I activate the caller waiting feature on the fxs's? So if phone 1 is being used, and I called phone 1 from phone 2, phone 1 will get call waiting tone, and from phone 2 will hear the connecting tones? I put callwaiting=yes in Zapata.conf already. But it didn't work.Any help? 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 ___ 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 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] IAX2 bug in DIAX solved - Great Thanks to Steven!
Hi, From: Andrew Thompson [EMAIL PROTECTED] Is this for the disconnection bug? Perhaps I need to flush diax and start fresh but it did the same for me. diax0.9.6d with new dll -- * -- diax0.9.6b with new dll Call lasted about 20-30 seconds. It is for no ring bug... Do you mean the call is disconnected after 20-30s? Can you provide more details? Thank you and best regards, Dan ___ 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] Outbound call with Go2Call
Any got experience with these? I couldn't fint anything in any postings... it seems they have a h.323 on voip01.go2call.com and a sip on sip01.go2call.com I have tried to register with some of the same as I use for nikotel, but Asterisk does not want to register. I've tried to use both the user name (ingvald) and the PIN code 440 as authentication. ---from sip.conf register = 440686267684:[EMAIL PROTECTED] [go2call] type=friend secret=X auth=md5 username=440686267684 authuser=ingvald fromuser=440686267684 host=sip01.go2call.com Any ideas, or where can I see all the options possible to pass to the server? -- With kind regards / Med vennlig hilsen Sjur Eivind Usken Hospitant i testnett gruppa Uninett AS +47 91772027 ___ 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] SNOM IAX image
There is no special IAX image. Just use SIP and it should work with Asterisk as well. CS -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Kannaiyan Natesan Sent: Monday, January 19, 2004 11:22 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] SNOM IAX image Is the SIP bin same for IAX as well? Kannaiyan - Original Message - From: Christian Stredicke [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, January 19, 2004 7:08 PM Subject: RE: [Asterisk-Users] SNOM IAX image For those who are using snom 200 phones, I think we have a promising image now ready at http://snom.com/download/share. Its version number is 2.03m. Please check this image; it should fix the known issues. The release notes can be found at http://www.snom.com/snom200_release_notes_de.php. If everything goes well, we will make also snom 100/105/220 images available. Christian -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Christian Stredicke Sent: Wednesday, January 14, 2004 11:08 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] SNOM IAX image Michael, There are a couple of images at http://snom.com/download/share. We are not really happy with the latest image yet; hopefully we can fix the remaining issues in a couple of days. Input appreciated (but no new feature requests until we have this stuff stable!). You to update the image: http://www.snom.com/faq/FAQ-02-08-31-cs.pdf. I guess if you have such a pretty old image, you should to a tftp update using the bootloader. Christian ___ 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 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 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] ultra-cheap asterisk box
--On Monday, January 19, 2004 12:25 PM +1100 [EMAIL PROTECTED] wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Albertson Sent: Friday, 16 January 2004 4:32 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] ultra-cheap asterisk box snip What I'm finding is that the PCs are so cheap that the cost of electric power to run them is now a large part of the cost. (assume 0.20/kwh times 200W times 365 days = $350. So you pay for the PC again every year in electric power to run it. Worse. In an office with airconditioning _all_ of that PC's 200W goes to heat and your A/C unit will use about 220W of power to remove that 200W of heat.) and at a small office they will not have a server room so noise from the fan is an issue. Are you sure the computer uses all the Power all the time? I would have thought that 200W was the peak, not the average. I guess the only way to measure it is to watch your home's power meter after you've turned off everything else :-) Radio Shack has a really neat A/C power meter that plugs into the wall. You then plug the A/C powered appliance you want to test into the unit. The unit reports instantaneous KW, KVA, and power consumption over time. It claims 15A max, but I've run it much higher. This is a great tool and a really fun gadget as well! It is one of those things you just don't want to spend the money on for a single test, but then when you have it you find uses for it constantly. I went around measuring everything in the house and at work after I got it. I think it was about $50, but on sale it was a good 30% off! ___ 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] New sounds also now in CVS
I keep noticing the references to words related to weather in this thread and I am getting more and more curious; why the weather related words for a PBX? What other broad topics for words exist right now besides those that are PBX specific and weather-related? ___ 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] DTMF with H.323
Hi All I have noticed a problem with dtmf reception on asterisk's side from H.323 clients (specifically clients sending in-band dtmf like NM). Asterisk v. 0.5.0 works perfectly while the latest release (0.7.1) never works. I am going ot look at the code later to see what has been changed Anybody else noticed this? Jason -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alexandru Coseru Sent: 20 January 2004 09:17 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] SS7 over Asterisk ? Now I understood that I can't use Asterisk for what I'm planning to do.. By using TDMoE , all I get is a new span on the destination server... But from there , I'm stuck... Anyway , thanks Regards Alex - Original Message - From: Philipp von Klitzing [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, January 19, 2004 8:07 PM Subject: Re: [Asterisk-Users] SS7 over Asterisk ? Hi! Maybe , I never tried TDMoE ... Where can I found a documentation or at least a sample for doing that ? http://www.asteriskdocs.org/current/docs-pdf/hgta.pdf page 29 Note that this book is still in pre-alpha state... Philipp ___ 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 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 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] ADSI phone vs. IP phone
On Mon, 19 Jan 2004, David Gomillion wrote: Andrew wrote: First, what's wrong with PoE? Is it any worse than installing tons of channel banks? Can anybody recommend a good PoE product? I am interested in getting that implemented. PJ -- Wisdom is not a product of schooling but the lifelong attempt to acquire it. -- Albert Einstein ___ 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] New sounds also now in CVS
On Tue, Jan 20, 2004 at 12:25:46AM -0800, Ken Alker wrote: What other broad topics for words exist right now besides those that are PBX specific and weather-related? I'd like prepaid calling phrases. PIN's, card numbers, account numbers, balance... ___ 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] Power Over Ethernet for *any* ethernet switch (or hub); product idea
Based on several threads I've read on this list, I assume that it would be handy to supply POE (power over ethernet) in an environment without having to purchase POE switches (assumed expensive) and abandon one's existing (familiar/custom/not-yet-expensed/etc.) switches/hubs. Assume I have a non-POE switch with 24 RJ-45 (ethernet) ports. I design a 1U box that can be mounted just above/below the non-POE switch, call it a POEI (POE inserter). This box has 48 RJ-45 ports, 24 inputs and 24 outputs. The end user removes all the ethernet cables connected to the existing switch and moves them to the outputs of the POEI. Next, the end user takes six-inch long ethernet cables and connects each (now vacant) port of the existing switch to the inputs of the POEI. The POEI simply connects the four ethernet signals on each of its inputs (pins 1,2,3,6 on each) to the same pins on its corresponding outputs. Additionally, it supplies -48VDC (maybe selectable if there are other voltage needs) on the appropriate pins (also maybe selectable if different vendors use different wiring conventions for POE) of its outputs. This could be an inexpensive way to provide POE without having to replace all of one's switches. Additionally, this could be a nifty business opportunity. Are POE switches expensive enough to warrant manufacturing above? If not, is there a case for not having to swap out all of ones existing switches? Does something like this already exist for cheap? If so, is it any good? If so, does it need more features? If not, would you buy something like this? If so, what features have I missed? If so, what is it worth? Daydreaming, as usual. Ken ___ 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] Power Over Ethernet for *any* ethernet switch (or hub); product idea
Ken Alker wrote: Based on several threads I've read on this list, I assume that it would be handy to supply POE (power over ethernet) in an environment without having to purchase POE switches (assumed expensive) and abandon one's existing (familiar/custom/not-yet-expensed/etc.) switches/hubs. Ken, such a device (and some more PoE stuff) is available from Powerdsine. Don't know what it costs, just wanted to let you know its available. Jan ___ 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] Power Over Ethernet for *any* ethernet switch (or hub); product idea
Froogle is your friend http://froogle.google.com/froogle?q=powerdsine - Original Message - From: Michiel Betel [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, January 20, 2004 3:08 AM Subject: Re: [Asterisk-Users] Power Over Ethernet for *any* ethernet switch (or hub); product idea Ken Alker wrote: Based on several threads I've read on this list, I assume that it would be handy to supply POE (power over ethernet) in an environment without having to purchase POE switches (assumed expensive) and abandon one's existing (familiar/custom/not-yet-expensed/etc.) switches/hubs. Assume I have a non-POE switch with 24 RJ-45 (ethernet) ports. I design a 1U box that can be mounted just above/below the non-POE switch, call it a POEI (POE inserter). This box has 48 RJ-45 ports, 24 inputs and 24 outputs. The end user removes all the ethernet cables connected to the existing switch and moves them to the outputs of the POEI. Next, the end user takes six-inch long ethernet cables and connects each (now vacant) port of the existing switch to the inputs of the POEI. Ken, The boxes youn describe are already being manufactured by amongst others: http://www.powerdsine.com/Products/Midspan/ I have no idea on pricing though.. Regards, Michiel ___ 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 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] New sounds also now in CVS
As a sugestion, store the sounds in a soundlib tree, hashed or categorised (boolean (yes, no, true,false, up, down etc.),numbers, caledar(day, date, time etc), state, weather etc) and dont duplicate any sounds then make a sounds tree with virtual categories and sim link to the files needed. This keeps the directory sizes down and allows for sound sets to be built up with all the words they use in them. It also allows sounds to be added as needed rather than requiring all sounds to be part of a distribution. Robert Hajime Lanning wrote: quote who=Tilghman Lesher Although the OS may cache that information, the userland process can take quite some time to process a very full directory. I've had this happen quite a few times with Linux ext2 filesystems, where the fileglob * exceeded bash's limit of 32,768 characters. /bin/ls on those directories took several minutes before the first results were given. I'll additionally comment that the directories I was working with were not normally that full, but was a side effect of a process dumping lots of little files into a directory when something went wrong. On a slight tangent, NT4 had a practical limit of about 300 directory entries before attempting to process the directory became unbearably slow. -Tilghman A couple of things, searching a directory for a specific name tends to be a linear search through the directory (unless the filesystem uses binary trees, like ReiserFS...), ls is a bad example of a command, it is more of a worse case example. ls will read the entire directory, sort it, then do a stat() on every file listed. All of this is done before it formats the output. So, you have to wait until it is all done, before you see the first character output. ___ 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] Remote reload Cisco 7960
You need a little more to make this script reboot the phone. It basically instructs the phone to check a file called 'syncinfo.xml' at its TFTP URL. This file needs to contain the following line: IMAGE VERSION=* SYNC=2/ The number 2 above is the sync value which must be different (I think higher) than the sync: field defined in your SIPDefault.cnf file. Then the script should do its stuff and reboot the phone. Rgds, Adam -Original Message- From: B. J. Bomar [mailto:[EMAIL PROTECTED] Sent: Monday, January 19, 2004 6:42 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Remote reload Cisco 7960 I've tried to use that script, but the phones seem to ignore it. I am in the process of upgrading to 6.1 on the phones, maybe they will behave like they're supposed to. B. J. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian West Sent: Friday, January 16, 2004 22:27 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Remote reload Cisco 7960 http://www.bkw.org/~brian/cisco/reboot7960.txt or you can us this handy perl script.. NEXT!!! bkw On Fri, 16 Jan 2004, Rich Adamson wrote: Does anyone have a working way of having a Cisco 7960 reload its config remotely. I have tried some of the scripts that I have found on the web, but to no avail. Thanks for the help. telnet to the box and reload it. command line has the ability. 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 ___ 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 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 * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise 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 ___ 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] open h323
Has anyone installed openh323 on redhat 9 ? I have problems compiling pwlib: In file included from /usr/include/openssl/ssl.h:179, from ../../ptclib/pssl.cxx:195: /usr/include/openssl/kssl.h:72:18: krb5.h: No such file or directory From what I see (google) there seems to be a general problem with pwlib, openssl and redhat 9. Can anyone help me out ? Regards, Dave ___ 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] Calls with incoming distinctive ring
Look into bugs.digium.com. I think there is a patch for doing what you want. - Original Message - From: Scott Bennett [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, January 19, 2004 11:01 PM Subject: RE: [Asterisk-Users] Calls with incoming distinctive ring So am I to assume this is not possible? Can someone let me know one way or another, or just at least flame me for asking? Hello List I have searched the lists, the wiki and the handbook and see how to use distinctive ring inside however I can't find incoming. I have 1 x100p and 2 phone numbers, My Voice calls are normal ring, my Fax are short short long. How do I tell * to route the call to an extension based on the ring candance? Is it possible? Right now it seems when the x100p sees the short short long it locks up and refuses to answer the line again. Thanks For Any Help You Can Provide! Scott ___ 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 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 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] open h323
Yes, I'm sorry that I can't remember the exact details. But there is a variable that you have to pass in with the make command to point to the include files associated with kerberos from memory. Look in the makefile for this. Something with a -I command in it :) Look for kerberos in the Makefile, maybe someone else knows it exactly. Then it just compiles a piece of cake. - Kim Has anyone installed openh323 on redhat 9 ? I have problems compiling pwlib: In file included from /usr/include/openssl/ssl.h:179, from ../../ptclib/pssl.cxx:195: /usr/include/openssl/kssl.h:72:18: krb5.h: No such file or directory From what I see (google) there seems to be a general problem with pwlib, openssl and redhat 9. Can anyone help me out ? Regards, Dave ___ 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 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] X100P CallOut Problems !!
Hi all, I just now receive the FXO X101P Card but can't at any way make then call out. I can hear the signal, even call but always receive from my local operator error that or the number don't exist or need more numbers. I play alot with txgain and rxgain, but none help me out. Being honest i try alot 5 hours and none !!! I'm using asterisk in his sample configs. I mean i call out using 1234 etc.. Zapata.conf is Ok Zaptel.conf is ok (I follow the Digium faqs, then for a good person that show-me this in the Asterisk IRC) ( Using here is an Asterisk 7.1) Did anyone know a txgain and rxgain from Brazilian lines ? (I'm trying with Vesper operator) Did i need make something more ( i know that need) :) Please could someone with lot's of time help-me out here with this simple question ? I just wanna call out too !!! Thanks alot ! ___ 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] Re: Power Over Ethernet for *any* ethernet switch (or hub); product idea
Jan Baumann [EMAIL PROTECTED] said: such a device (and some more PoE stuff) is available from Powerdsine. Don't know what it costs, just wanted to let you know its available. It's of the expensive-SNMP-managed-kit kind :-). Wouldn't a 19 RJ45 strip, a bit of cable to wire the thing up and a COTS powersupply do the trick? Only issue of course is that with 24 ports, you'd need a quality powersupply. Anyway, such strip seems to be DIY'ed together already by some: http://www.nycwireless.net/poe/ at the bottom of the page. -- Cees de Groot http://www.tric.nl [EMAIL PROTECTED] tric, the new way helpdesk/ticketing software, VoIP/CTI, web applications, custom development ___ 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] unsubscribe
unsubscribe ___ 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] Codec matching weirdness
Hi! A better option and one Asterisk desperately needs is some kind of --lint option, Which would check the config for errors and useless misspelled options. smile I personal find one or more typos or misspelling a month, On my PBXs. Yes, indeed, same for me. My advice is to always do an extensions reload and immediately check the /var/log/asterisk/messages, but that'll still not catch everything. Just found this which explained strange errors I saw for two months: exten = 123,4,Playback(some-sound)) Haha - stupid, ain't it? Cheers, Philipp ___ 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] unsubscribe
Sam wrote: unsubscribe ___ 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 Could it be any clearer.. Later.. ___ 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] RE: Latest version of asterisk
Nope, it's the T400P, the old one that they don't sell anymore. I actually haven't seen any issues with it and RH 9. it seems to run just fine. MATT--- -Original Message- From: Aram Ter-Martirosyan [mailto:[EMAIL PROTECTED] Sent: Monday, January 19, 2004 11:42 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] RE: Latest version of asterisk Hello Matt, Is that the Wildcard TE410P you are using. Digium said that it had some problems with Redhat 9.0 is that correct? - Digium quad T1 card - 3 T1's (2 x B8ZS ESF Long Distance and 1 x robbed-bit SF local) - Redhat 9.0 Aram Ter-Martirosyan Senior Account Manager Hi-Tech Gateway, Inc. http://www.hi-teck.com 1225 Grand Central Ave. Glendale, CA 91201 [EMAIL PROTECTED] tel 818.546.4601 fax 818.546.4617 Turning Technology Into Business Solutions -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of mattf Sent: Monday, January 19, 2004 6:21 PM To: '[EMAIL PROTECTED]' Subject: RE: [Asterisk-Users] RE: Latest version of asterisk Hello, Our max for a single machine is 40 concurrent SIP - Zap conversations for about a 12 hour period and over 5000 total phone calls per day. We didn't see crashes going over that, but we wanted to be safe and now have 2 identical machines handling upto about 30 concurrent SIP - Zap calls(3000 phone calls per day), and a third old machine for office use that never gets over 10 concurrent calls. Here's the specs for these systems: - 120 installed hardphones: - 80 x grandstream 102 hardphones - 20 x Sipura analog adapters(2 phones each) - 2 x Asterisk servers - 2.6 GHz Pentium4 800MHz bus w/ HyperThreading enabled - Asus p4c800 800MHz mobo - 2GB DDR400 RAM (This is actually overkill you need 1GB max if you reboot weekly) - 4 x 36GB SCSI drives in RAID 10 w/megaraid card - 3com 905CX ethernet card - Digium quad T1 card - 3 T1's (2 x B8ZS ESF Long Distance and 1 x robbed-bit SF local) - Redhat 9.0 - Asterisk with many modules turned off and no MOH With these servers you can see the load average jump from 0.00 to 6.25 in a matter of a minute and then back down again, all while never dropping a call or crashing. We also recently diagnosed our lock-freeze to the touchy manager interface(if you are logged into the manager interface and you loose connection, the manager outgoing buffer seems to overflow and freeze Asterisk). So it doesn't seem to be a problem of hardware. But we still haven't figured out how to fix it. One note as to Ethernet cards, we actually fried a Realtek 8139 Ethernet card that we had put in a server temporarily as we were doing our testing. It started to generate a lot of errors and dropping packets left and right. When we took it out it was VERY hot. We then put in a 3com 905 card and haven't had an issue with it yet. Hope this helps, MATT--- -Original Message- From: T. Chan [mailto:[EMAIL PROTECTED] Sent: Monday, January 19, 2004 4:49 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] RE: Latest version of asterisk Thanks, Matt ! So, am I correct in assuming that there are quite a few (or alot) of us who have had not so good experiences with Asterisk? That Asterisk would crash after it hit a certain number of calls or after a certain period of time with 15-20 calls? I understand that there were others who were able to send a good number of calls through but can anyone tell us if they have had tested and confirmed that Asterisk runs better without or with HT and in terms of number of calls, how many would each one support, in the ballpark? It would also be nice if one could tell us the computer configuration in order to send that many calls without crashing Asterisk. Does it make a difference running the LAN on a ONBOARD LAN card as compared to a PCI Intel or 3COM LAN card, since there is a chance that packets are passing more efficiently on a PCI LAN card? Side question: Is it possible to do passthrough faxing? Like, customers sending me H323 or SIP fax calls and the Asterisk will pass through to another gateway? Anyone successful in doing that? Tommy -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of mattf Sent: Monday, January 19, 2004 8:32 AM To: '[EMAIL PROTECTED]' Subject: RE: [Asterisk-Users] RE: Latest version of asterisk Hello, I've had Asterisk installed on HT capable machines in both HT mode(with SMP) and non HT mode (with non-SMP) and did not notice any differences functionally between them. The processor load was always less in HT SMP mode than non HT and I have experienced Asterisk deadlocks in both modes so it doesn't really seem to matter if you leave HT on(at least in my experiences). HT basically works by splitting off commands to one of two different virtual processors that both run at about 70% of processor's speed(that's why you may notice compiling to take longer
RE: [Asterisk-Users] R2 support
Ok, it's old and clunky, but in some countries like Brazil, Argentina and China is the only alternative. So, the original question, does anybody know something about the Steve's project or know a release date? All the best, Pablo. -Original Message- From: Alfred R. Nurnberger [mailto:[EMAIL PROTECTED] Posted At: Tuesday, January 20, 2004 1:11 Posted To: Asterisk Conversation: [Asterisk-Users] R2 support Subject: RE: [Asterisk-Users] R2 support Steve. You are saying this from your view of 2004. But at the time R2 was developed there were no microcontrollers and tones were decoded with LC filters. R2 provides interactive capabilities base on a simple tones protocol to retrieve ANI, dialed numbers, signalling status etc. It's compelled structure provides some kind of handshaking to deal with different kind of switches and their speed. Nowadays this is no issue at all but at the days R2 was developed you had to take into account that relays and step by step switches take their time. On the other hand I have to agree with you... Well, what is the definition of sane anyway :-) Regards. Alfred. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Steve Underwood Sent: Monday, January 19, 2004 5:08 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] R2 support Olle E. Johansson wrote: LQ (Asterisk) wrote: Hi guys, I was reading that Steve Underwood is working on Asterisk R2 signalling support, and has the 95% of the work done. What is R2? I'm curious. Half of R2D2, of course. Its also a stupid clunky multi-tone based telephone signaling system widely used over E1s in South America, Asia, and parts of Eastern Europe. No sane telecoms engineer would use it. However, few telecoms engineers are entirely sane :-) Regards, 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 ___ 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 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] Still problems at compiling
Hello experts, to avoid any unknown problems with my Linux installation I have now as a last resort method installed SuSE Linux 9.0 a new and have downloaded a fresh copy of Asterisk via CVS. Then I followed the steps of the Getting started with Asterisk and compiled successfully zaptel and libpri (as far as I can see). But when I compile asterisk I get an error. I have attached the sysout log below. Any hint and help highly appreciated. What is wrong. Franz the sysout log during make clean and make install of asterisk - linux:/usr/src/asterisk # make clean for x in res channels pbx apps codecs formats agi cdr astman stdtime; do make -C $x clean || exit 1 ; done make[1]: Entering directory `/usr/src/asterisk/res' rm -f *.so *.o .depend make[1]: Leaving directory `/usr/src/asterisk/res' make[1]: Entering directory `/usr/src/asterisk/channels' rm -f *.so *.o .depend rm -f busy.h ringtone.h gentone gentone-ulaw make[1]: Leaving directory `/usr/src/asterisk/channels' make[1]: Entering directory `/usr/src/asterisk/pbx' rm -f *.so *.o .depend make[1]: Leaving directory `/usr/src/asterisk/pbx' make[1]: Entering directory `/usr/src/asterisk/apps' rm -f *.so *.o look .depend make[1]: Leaving directory `/usr/src/asterisk/apps' make[1]: Entering directory `/usr/src/asterisk/codecs' rm -f *.so *.o .depend ! [ -d g723.1 ] || make -C g723.1 clean ! [ -d g723.1b ] || make -C g723.1b clean make -C gsm clean make[2]: Entering directory `/usr/src/asterisk/codecs/gsm' rm -f */*.o\ ./tst/lin2cod ./tst/lin2txt \ ./tst/cod2lin ./tst/cod2txt \ ./tst/gsm2cod \ ./tst/*.*.* find . \( -name core -o -name foo \) \ -print | xargs rm -f rm -f ./lib/libgsm.a ./add-test/add \ ./bin/toast ./bin/tcat ./bin/untoast\ ./gsm-1.0.tar.Z make[2]: Leaving directory `/usr/src/asterisk/codecs/gsm' make -C lpc10 clean make[2]: Entering directory `/usr/src/asterisk/codecs/lpc10' rm -f *.o ./liblpc10.a make[2]: Leaving directory `/usr/src/asterisk/codecs/lpc10' make -C ilbc clean make[2]: Entering directory `/usr/src/asterisk/codecs/ilbc' rm -f libilbc.a *.o make[2]: Leaving directory `/usr/src/asterisk/codecs/ilbc' make[1]: Leaving directory `/usr/src/asterisk/codecs' make[1]: Entering directory `/usr/src/asterisk/formats' rm -f *.so *.o .depend make[1]: Leaving directory `/usr/src/asterisk/formats' make[1]: Entering directory `/usr/src/asterisk/agi' rm -f *.so *.o look .depend make[1]: Leaving directory `/usr/src/asterisk/agi' make[1]: Entering directory `/usr/src/asterisk/cdr' rm -f *.so *.o .depend make[1]: Leaving directory `/usr/src/asterisk/cdr' make[1]: Entering directory `/usr/src/asterisk/astman' rm -f *.o astman .depend make[1]: Leaving directory `/usr/src/asterisk/astman' make[1]: Entering directory `/usr/src/asterisk/stdtime' rm -f libtime.a *.o test .depend make[1]: Leaving directory `/usr/src/asterisk/stdtime' rm -f *.o *.so asterisk .depend rm -f build.h rm -f ast_expr.c make -C db1-ast clean make[1]: Entering directory `/usr/src/asterisk/db1-ast' rm -f libdb1.a libdb.so.2 hash.o hash_bigkey.o hash_buf.o hash_func.o hash_log2.o hash_page.o ndbm.o bt_close.o bt_conv.o bt_debug.o bt_delete.o bt_get.o bt_open.o bt_overflow.o bt_page.o bt_put.o bt_search.o bt_seq.o bt_split.o bt_utils.o db.o mpool.o rec_close.o rec_delete.o rec_get.o rec_open.o rec_put.o rec_search.o rec_seq.o rec_utils.o hash.os hash_bigkey.os hash_buf.os hash_func.os hash_log2.os hash_page.os ndbm.os bt_close.os bt_conv.os bt_debug.os bt_delete.os bt_get.os bt_open.os bt_overflow.os bt_page.os bt_put.os bt_search.os bt_seq.os bt_split.os bt_utils.os db.os mpool.os rec_close.os rec_delete.os rec_get.os rec_open.os rec_put.os rec_search.os rec_seq.os rec_utils.os make[1]: Leaving directory `/usr/src/asterisk/db1-ast' make -C stdtime clean make[1]: Entering directory `/usr/src/asterisk/stdtime' rm -f libtime.a *.o test .depend make[1]: Leaving directory `/usr/src/asterisk/stdtime' linux:/usr/src/asterisk # make install ./mkdep -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\CVS-01/20/04-10:14:14\ -DINSTALL_PREFIX=\\ -DASTETCDIR=\/etc/asterisk\ -DASTLIBDIR=\/usr/lib/asterisk\ -DASTVARLIBDIR=\/var/lib/asterisk\ -DASTVARRUNDIR=\/var/run\ -DASTSPOOLDIR=\/var/spool/asterisk\ -DASTLOGDIR=\/var/log/asterisk\ -DASTCONFPATH=\/etc/asterisk/asterisk.conf\ -DASTMODDIR=\/usr/lib/asterisk/modules\ -DASTAGIDIR=\/var/lib/asterisk/agi-bin\ -DBUSYDETECT_MARTIN -DNEW_PRI_HANGUP `ls *.c` cli.c:31:19: build.h: No such file or directory dlfcn.c:40:25: mach-o/dyld.h: No such file or directory dlfcn.c:41:26: mach-o/nlist.h: No such file or directory dlfcn.c:42:28: mach-o/getsect.h: No such file or directory for x in res channels pbx apps codecs formats agi cdr astman
RE: [Asterisk-Users] ADSI phone vs. IP phone (and proper implementation thereof)
Probably because it's well known that these setups are prone to failure of either the PC's connection, the phone's connection, or degredation of one/both. It also breaks switch envirenments where spanning-tree portfast is enabled (not as big of a deal if the deployment is in concert with the infrastructure group, as it should be). Vendors should NEVER have implemented this functionality into phones unless it was working under all conditions. Personal experience shows that it is most definitely not on Cisco and 3Com products. Others have told me their stories with other manufacturer's equipment. None of it was good. It's not a production-stable way to deploy phones. Period. I'm wondering if what you say is actually true. According to recent media releases, Cisco has shipped over 2 million of their IP phones. They must be doing something right. Their phones are _designed_ to function and cooperate with the switch. Obviously, the installer has to be totally familiar with all phone, switch, router and network settings in order to have a successful installation. The switch needs to be configured with specific port, vlan, and class of service settings. Accepted practice is to provide a voice vlan and a data vlan. On the phone side, the phone knows to send voice on the specific vlan told to it by the switch , and to pass through data from the pc through the vlan told to it by the switch. The phone knows to prioritize voice traffic over data traffic. So does the switch. And so on through the connection of switches and routers. This ensures voice quality and precedence through out the network. Voip quality is not necessarily about bandwidth (because it works on T1 data lines as well as GB ports), but about instantaneous bottlenecks in the network. These instantaneous and random bottlenecks can occur in the cad environment mentioned. But with appropriate COS (layer 2) and TOS (layer 3) settings in the phones, switches, and routers, these bottlenecks become non- issues. In addition, what many people forget, or learn by experience, is that you absolutely _must_ have everything running full-duplex, and to physically check errors and statistics on each port of the switch in order to verify that you have error free links. You won't believe how many networks out there are broken because noone checks and fixes these issues. A voip network _must_ have managed switches so you can verify these things. There was mention of a heavy cad environment. Say your computer is connected to the 100mbps port of the phone. A g.711 call comes through. The call takes around 80 kbps. If I've done the math properly, the voice call takes only 0.08% of the bandwidth, hardly something that will interfere with 'heavy cad users'. More likely the opposite, the heavy cad users will interfere with the call, _but_ _only_ if the switch and phone are not configured properly for vlan, cos, tos, speed, and duplex settings. So having said this, you mentioned that you have had personal experience where this functionality is built into, or does not work in Cisco's case. Couldn't agree with you more (and I'm not the original poster). We've spent a number of years conducting independent (no vendor alliances) network performance assessments for corporations in more then 40 states, and have found a large percentage of network managers and technicians just don't pay attention to these things (for lots of reasons). As far as the switch function built into many of the sip phones, there has been a fair number of folks on this list that have had problems with it. If I recall correctly, John Todd (very experienced) recently queried the list relative to unusual C7960 switch problems. Unknown as to whether the root-cause was hardware failures, STP, or what, but maybe John will post his findings. Given our extensive experience with performance analysis, I would not use the switch function if it was limited to 10 meg half duplex except in very low usage office environments. It would be very interesting to hear from those that have real life experience with the switch function in network environments that are much larger then the typical SOHO shops, and that have invested the time/effort to properly diagnose the real root-cause of such issues. 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
Re: [Asterisk-Users] ADSI phone vs. IP phone
On Tue, Jan 20, 2004 at 02:08:33PM -0800, PJ said: On Mon, 19 Jan 2004, David Gomillion wrote: Andrew wrote: First, what's wrong with PoE? Is it any worse than installing tons of channel banks? Can anybody recommend a good PoE product? I am interested in getting that implemented. Several models of Cisco switches have PoE. Combined with VLAN trunking, they work well. If your networking gear is getting tired, a VoIP roll out is a good time to update your network infrastructure. ___ 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] Re-Invite between SIP phones
Anybody knows what do I need to tell Asterisk to issue a re-INVITE between two SIP phone to avoid having the media going through the server? Tks, Al __ Do you Yahoo!? Yahoo! Hotjobs: Enter the Signing Bonus Sweepstakes http://hotjobs.sweepstakes.yahoo.com/signingbonus ___ 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] Still problems at compiling
Franz Edler wrote: Hello experts, to avoid any unknown problems with my Linux installation I have now as a last resort method installed SuSE Linux 9.0 a new and have downloaded a fresh copy of Asterisk via CVS. IIRC there have been many who have tried and failed to build Asterisk on SuSE.. Have you tried to install it on RH9, I have never had a problem with RH9.. and apparently Fedora Core 1 is also working well.. If you want an RH9 install guide you can look at.. http://members.lycos.co.uk/wipe_out/asterisk Later.. ___ 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] Re-Invite between SIP phones
canreinvite=yes within sip.conf entities ... -Original Message- From: Al [mailto:[EMAIL PROTECTED] Sent: Tuesday, January 20, 2004 2:06 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Re-Invite between SIP phones Anybody knows what do I need to tell Asterisk to issue a re-INVITE between two SIP phone to avoid having the media going through the server? Tks, Al __ Do you Yahoo!? Yahoo! Hotjobs: Enter the Signing Bonus Sweepstakes http://hotjobs.sweepstakes.yahoo.com/signingbonus ___ 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 * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise 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 ___ 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] Still problems at compiling
On Tue, 2004-01-20 at 13:27, Franz Edler wrote: configure: error: termcap support not found make: *** [editline/libedit.a] Error 1 linux:/usr/src/asterisk # Did you actually read the error message and try to understand solve the problem? The very first answer from google gives you the exact same question and an answer: http://www.google.nl/search?q=site%3Alists.digium.com+configure%3A+error%3A+termcap+support+not+foundie=UTF-8oe=UTF-8hl=nlbtnG=Google+zoekenlr= Hint: install termcap devel package (and all other prerequisite devel packages in case you get more errors). Patrick ___ 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] SIP: Register that isn't a register?
Hi! WARNING[8201]: chan_sip.c:4821 handle_response: Got 200 OK on REGISTER that isn't a register This is most probably cause by registration of * with FWD. I am seeing this with iptel.org -Walter I had this when registering to FWD from * inside my LAN and without externip configured, If * sends its internal IP, the FWD server returns this message. Hm... in my case * has a public IP and is not behind NAT. It is, however, protected by the central university router/firewall... Anyway, I also see that on a 2nd machine that has a dynamic IP on a cable modem (also not behind NAT). So there must be more to it. Cheers, Philipp ___ 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] Broken macros during transferring call
Hi,
I have some issues with the use of macros when dialling. I use a macro,
similar to the stdexten macro to dial extensions. When I use the
astman program to transfer the recipient of a call made via the macro
to meetme for example it appears as if control is transferred into the
start state in the context of the caller. Hence the the extension that
you are trying to transfer to is lost and the transfer fails. Well.. actually,
it's failing today like that when transferring a call made to an iaxclient
located at a foreign iax server. Yesterday when I was testing to sip
clients located at a foreign iax server it appears as if control was passed
to state 2 with the extention in the context of the caller. The second case
I was able to work around, the first case is not easy as I seem to loose
the extension that one is trying to transfer to.
Any clues as to the correct approach to solve this? Yesterday I solved it
by making sure that all calls are made by macros or gotos that never return
and then adding exten = _.,2,Goto(${CONTEXT},${EXTEN},1) into
the default context, however if it returns into the default context with the
start state like it does when I transfer a call made to a foreign iaxclient
I am unable to fix this.
- Kim Hendrikse
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RE: [Asterisk-Users] CVS Changes (NAT-SIP)
It is not working. Need HELP -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian West Posted At: Tuesday, January 20, 2004 1:08 AM Posted To: Asterisk User Group Conversation: [Asterisk-Users] CVS Changes (NAT-SIP) Subject: Re: [Asterisk-Users] CVS Changes (NAT-SIP) Can you clarify this? Does it or doesn't it work? bkw On Mon, 19 Jan 2004, Asterisk User Group wrote: I had been running an older patched CVS to get VOIP working with NAT and everything had been running fine. I just built * on a new box with CVS-01/18/04-12:19:25. And now I can get remote SIP users to register. Has anything major changed... [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to externip = 69.132.68.17 ; Address that we're going to put in SIP messages if we're behind a NAT localnet = 192.168.1.0 ; Internal NETWORK address localmask = 255.255.255.0 ; Internal netmask context = default ; Default for incoming calls ;srvlookup = yes; Enable SRV lookups on outbound calls ;pedantic = yes ; Enable slow, pedantic checking for Pingtel ;tos=lowdelay ;tos=184 ;maxexpirey=3600; Max length of incoming registration we allow ;defaultexpirey=120 ; Default length of incoming/outoing registration ;notifymimetype=text/plain ; Allow overriding of mime type in NOTIFY ;videosupport=yes ; Turn on support for SIP video disallow=all; Disallow all codecs allow=ulaw ; Allow codecs in order of preference allow=ilbc [1001] type=friend secret=1001 host=dynamic username=1001 mailbox=1001 context=local nat=no [1006] type=friend secret=oicu812 host=dynamic username=1006 mailbox=1006 context=local nat=yes canreinvite=no qualify=500 Internal SIP users can register it just the outside users. -gcc ___ 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 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 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] AG4000C and T100P
Hi, Im currently working on a * server connected to an ahem Wireless Communication Server. The HWCS has an NMS AG 4000C. Im using NI2, net side on the * box. The D-Channel comes up. The B-Channels come up. The first call to the * box goes through. After which, the HWCS with the AG4000 seems to either get a state-machine screwed up, or * does not send the correct hangup sequence from the first call, because directly after the connect on the next call, the HWCS sends a hangup, but keeps the channel to the handset open. Most of the time, I only hear silence, but sometimes I hear white noise. Loud white noise. Here is an annotated dump: ( By the way, this is using a hacked libpri that does not send channel identification during the ALERTING and othersuch redundant information, and yes, the behaviour was EXACTLY the same before I hacked the lib ) Sorry, had to cut the dump due to size restrictions, but is available upon request -- Mike Dexter Church
[Asterisk-Users] SIP: outbound calls
Hi all, Any advice on how to place a call from a SIP UA routed through *? Do I just place a sip call to [EMAIL PROTECTED]:5060 ? I am a little confused, since all of my Uas require registration for presence information. Thanks in advance, Tim -- Notice: This e-mail message, together with any attachments, contains information of Merck Co., Inc. (One Merck Drive, Whitehouse Station, New Jersey, USA 08889), and/or its affiliates (which may be known outside the United States as Merck Frosst, Merck Sharp Dohme or MSD and in Japan as Banyu) that may be confidential, proprietary copyrighted and/or legally privileged. It is intended solely for the use of the individual or entity named on this message. If you are not the intended recipient, and have received this message in error, please notify us immediately by reply e-mail and then delete it from your system. -- ___ 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] Re-Invite between SIP phones
I'd suggest placing a packet sniffer (tcpdump, etherreal) and see whats happening because it works great for me and always has but I guess it also requires support on the end-points and possibly (assuming non-cisco enviro) there maybe an option that needs to be configured on your phones/gateways. Please provide more information on your setup ... -Original Message- From: Al [mailto:[EMAIL PROTECTED] Sent: Tuesday, January 20, 2004 2:52 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Re-Invite between SIP phones Already did that, but it's not working. Al --- Low, Adam [EMAIL PROTECTED] wrote: canreinvite=yes within sip.conf entities ... -Original Message- From: Al [mailto:[EMAIL PROTECTED] Sent: Tuesday, January 20, 2004 2:06 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Re-Invite between SIP phones Anybody knows what do I need to tell Asterisk to issue a re-INVITE between two SIP phone to avoid having the media going through the server? Tks, Al __ Do you Yahoo!? Yahoo! Hotjobs: Enter the Signing Bonus Sweepstakes http://hotjobs.sweepstakes.yahoo.com/signingbonus ___ 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 * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise 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 ___ 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 __ Do you Yahoo!? Yahoo! Hotjobs: Enter the Signing Bonus Sweepstakes http://hotjobs.sweepstakes.yahoo.com/signingbonus ___ 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 * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise 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 ___ 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] Power Over Ethernet for *any* ethernet switch (or hub); product idea
Hi. The POEI simply connects the four ethernet signals on each of its inputs (pins 1,2,3,6 on each) to the same pins on its corresponding outputs. Additionally, it supplies -48VDC (maybe selectable if there are other voltage needs) on the appropriate pins (also maybe selectable if different vendors use different wiring conventions for POE) of its outputs. and probably you're going to fry something on your lan. POE isn't simple power on the right pins, but is a sort of protocol. Really, on POE enabled devices (or injectors) you won't measure the DC with a tester, simply because POE on port X is enabled after a request by the device on that port. this is for mantaining compatibity with non POE devices. so you will need also something that detects the power request on each port and enables it. Matteo. -- Matteo Brancaleoni Espia System Administrator Email : [EMAIL PROTECTED] Web : http://www.espia.it Phone : +39 02 70633354 - ext 201 IAX(2): [EMAIL PROTECTED] - ext 201 Iaxtel: 1-700-56-62458 - ext 201 ___ 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] Re-Invite between SIP phones
I'm trying to place calls between Cisco ATAs and XLite clients. Calls go through perfectly. Both sides of the call negotiate the same CODEC (G711a). I read that older Cisco ATA 186 firmwares don't support reinvites but when capturing traffic there is no Asterisk attempt to send the reinvite message. Al --- Low, Adam [EMAIL PROTECTED] wrote: I'd suggest placing a packet sniffer (tcpdump, etherreal) and see whats happening because it works great for me and always has but I guess it also requires support on the end-points and possibly (assuming non-cisco enviro) there maybe an option that needs to be configured on your phones/gateways. Please provide more information on your setup ... -Original Message- From: Al [mailto:[EMAIL PROTECTED] Sent: Tuesday, January 20, 2004 2:52 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Re-Invite between SIP phones Already did that, but it's not working. Al --- Low, Adam [EMAIL PROTECTED] wrote: canreinvite=yes within sip.conf entities ... -Original Message- From: Al [mailto:[EMAIL PROTECTED] Sent: Tuesday, January 20, 2004 2:06 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Re-Invite between SIP phones Anybody knows what do I need to tell Asterisk to issue a re-INVITE between two SIP phone to avoid having the media going through the server? Tks, Al __ Do you Yahoo!? Yahoo! Hotjobs: Enter the Signing Bonus Sweepstakes http://hotjobs.sweepstakes.yahoo.com/signingbonus ___ 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 * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise 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 ___ 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 __ Do you Yahoo!? Yahoo! Hotjobs: Enter the Signing Bonus Sweepstakes http://hotjobs.sweepstakes.yahoo.com/signingbonus ___ 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 * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise 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 ___ 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 __ Do you Yahoo!? Yahoo! Hotjobs: Enter the Signing Bonus Sweepstakes http://hotjobs.sweepstakes.yahoo.com/signingbonus ___ 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] Re-Invite between SIP phones
Hi, I think canreinvite=yes won't work in most of the situations. I have implemented Redirect SIP 300 Message to redirect to the SIP address you speficy in the sip.conf. Where you can have , register = username:[EMAIL PROTECTED]/extension [extension] redirect=yes redirecturi=sip:[EMAIL PROTECTED] redirecturi=sip:[EMAIL PROTECTED] ... will make to redirect to all the URI's yu specify in the sip.conf. I'm also working on this so that it can get the redirections from the database rather than reloading asterisk all the time when you modify the redirection uri. You can check through that. http://bugs.digium.com/bug_view_page.php?bug_id=879 Message transmission is alright, but for some reason it is not working. Can you test with yours and let me know where is the problem, I will modify the code once you get the clue where is the problem on it. If successfully please send me the sip debug message and I will just make sure it works for all. Kannaiyan - Original Message - From: Low, Adam [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, January 20, 2004 2:00 PM Subject: RE: [Asterisk-Users] Re-Invite between SIP phones I'd suggest placing a packet sniffer (tcpdump, etherreal) and see whats happening because it works great for me and always has but I guess it also requires support on the end-points and possibly (assuming non-cisco enviro) there maybe an option that needs to be configured on your phones/gateways. Please provide more information on your setup ... -Original Message- From: Al [mailto:[EMAIL PROTECTED] Sent: Tuesday, January 20, 2004 2:52 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Re-Invite between SIP phones Already did that, but it's not working. Al --- Low, Adam [EMAIL PROTECTED] wrote: canreinvite=yes within sip.conf entities ... -Original Message- From: Al [mailto:[EMAIL PROTECTED] Sent: Tuesday, January 20, 2004 2:06 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Re-Invite between SIP phones Anybody knows what do I need to tell Asterisk to issue a re-INVITE between two SIP phone to avoid having the media going through the server? Tks, Al __ Do you Yahoo!? Yahoo! Hotjobs: Enter the Signing Bonus Sweepstakes http://hotjobs.sweepstakes.yahoo.com/signingbonus ___ 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 * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise 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 ___ 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 __ Do you Yahoo!? Yahoo! Hotjobs: Enter the Signing Bonus Sweepstakes http://hotjobs.sweepstakes.yahoo.com/signingbonus ___ 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 * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise 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 ___ 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 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] Compiling problems with SuSE
-Original Message- From: Uwe Klein [mailto:[EMAIL PROTECTED] Sent: Monday, January 19, 2004 9:14 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Compiling problems with SuSE From: Dustin Knuttgen on Sunday, January 18, 2004 11:47 PM We tried to use SuSE initially and had no luck compiling zaptel on either 8.2 or 9.0. We even had Digium take a look. After working on it for days we finally switched to Red Hat 9. Is there anyone who succeeded in compiling Asterisk with SuSE 8.2 or 9.0? HI Dustin, what kind of error did you get? something like this: pbx.c:581: warning: comparison between signed and unsigned pbx.c: In function `pbx_substitute_variables_temp': pbx.c:765: warning: comparison between signed and unsigned pbx.c:812: warning: comparison between signed and unsigned pbx.c: In function `pbx_builtin_hangup': pbx.c:4017: internal compiler error: Segmentation fault ?? I had problems with SuSE 8.2 and Asterisk from cvs dated ~12July2003 I got it fixed by adding 128MB of memory to the 32MB on this P200 machine. with 300MB of swap it should not have made a difference ( except taking forever ) but it did. G! UK -- Uwe Klein [mailto:[EMAIL PROTECTED] KLEIN MESSGERAETE Habertwedt 1 D-24376 Groedersby b. Kappeln, GERMANY phone: +49 4642 920 123 FAX: +49 4642 920 125 ___ 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 Uwe, I had a problem at the end when it does the depmod -a. We got an error with around ten modules. The only thing I could find related to the errors was something about PPP in the kernel or in the Makefile. Neither of which made any difference. Dustin ___ 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] Still problems at compiling
From: Patrick Sent: Tuesday, January 20, 2004 2:29 PM Did you actually read the error message and try to understand solve the problem? No, being a Linux newbee and under a stress condition, I did not. But meanwhile I did and I installed several additional packages and now the compilation came to an end and brought an executable asterisk code. There were also various warnings during compilation which I generously ignored for this time Thanks for your patience with a stressed newbee. Patrick ___ 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 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] [ot] Grandstream hardware
On Tue, 2004-01-20 at 01:12, Nicolas Bougues wrote: A DSP is a processor. Just like when you buy a Pentium IV, it doesn't give you the right to use, for instance, MS Windows on it. You have to pay for software. And that's what algorithms are. Except that you have to pay for algorithms even if you do your own original implementation. Yes, but with a Pentium you don't have to pay a license to use MMX in your software, since the MMX instructions are part of the product you are allowed to use them with that product. If I understand things correctly, the companies that make DSP chips can implement whatever codec(s) they want and NOT have to pay the patent holders to sell this product with the patent holder's codec in it? I ask again, how does Grandstream (from all accounts a very small company) afford to provide the patented codecs in their products? --Eric ___ 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] wink time
Hi list, I have an X100P to place some outgoing calls. But sometimes zttool shows a red alarm and after I unplug and plug the line cable, the alarm is cleared. Sometimes dialing works and sometimes not. I suspect it's a timing problem. Could someone point me on how to configure timing parameters for an X100P? thanks in advance Eduardo ___ 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] [ot] Grandstream hardware
If I understand things correctly, the companies that make DSP chips can implement whatever codec(s) they want and NOT have to pay the patent holders to sell this product with the patent holder's codec in it? That is not true. You must license any technologies you use if their license demands it. I ask again, how does Grandstream (from all accounts a very small company) afford to provide the patented codecs in their products? Volume? An excellent sales contract? Perhaps the DSP or DSP firmware they bought to aid their development has licenses for the commercial codecs present? There are a number of MP3 decoder ICs which include the MP3 license cost in the cost of the chip itself, for example. Regards, Andrew ___ 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] configuration to Grandstream via tftp
Replying to myself. The GS phones use TFTP extensions (RFC 2347) to
provide additional info in their TFTP requests. The server has to be
aware of these extensions, if it wants to serve different files.
Here is a small dump for a request from an HandyTone (key/value) :
grandstream_MODELHT-100
grandstream_NAT1
grandstream_ID000b8200c14a
grandstream_REV_BOOT00100013
grandstream_REV_PHONE00104026
grandstream_REV_VOC0012
grandstream_REV_HTML00100020
grandstream_REV_VP0010
We can easily see the MAC address, the model and the current firmware
versions (1.4.26).
With these informations, the TFTP server could :
- serve the right cfg.txt file
- serve the right firmware files (or actually, serving nothing if the
server considers the phone to be up to date).
I'll try to see if my basic Java knowledge enables me to make the
NAT-aware TFTP server fwtftpd understand these extensions.
Hi Nicolas,
I've also tried to hack a little with the fwtftpd today to serve the cfg.txt
to the phones (and also updates to software). I cannot get it to accept the
cfg.txt i give it though - have anyone successfully served that file to
their GS phone? or made it update the firmware with fwtftpd?
I made a updated fwftpd.java file - anyone is welcome to test it here:
http://musimi.dk/fwtftpd.java
it uses the mac address of the GS phone and then sends the /cfg/{MAC}.txt
file when the phone requests the cfg.txt
Please help with getting the phone to accept the config file? - should i
send something back as OptionsACK to show the GS phone that it is ok to
update?
Cheers,
Jens Davidsen
Musimi.dk
--
Nicolas Bougues
Axialys Interactive
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Re: [Asterisk-Users] WANTED: Toll-Free gateways in Europe/Asia/Africa/South America
Looks like the list server is really lagging tonight. I found out some more info so will just post it in a new email with the same subject. I added: search = freenum.org to enum.conf and got a match (SIP system) when doing the lookup Maybe I overlooked that in the original instructions. Now will work on trying to get only IAX responses since SIP is rather problematic from behind the NAT router. IAX should work fine. John, Thanks for the tips on debugging. It pointed me in the right direction. Robert Robert - IAX as a protocol is completely dependent on the far-end gateway, and not on any specifications you can change. All the gateways at the moment only support SIP; none support IAX or IAX2, though hopefully that will change since some of them are actually running Asterisk as the media gateway. As soon as they offer IAX in addition to SIP, then we'll also need to re-examine the way that Asterisk handles ENUM lookups since currently only one NAPTR is handed back to the dialplan. For those nations that have multiple gateways or providers, I have put all the entries in a round-robin fashion so that the answers will be rotated by most standard DNS resolver libraries. However, this quickly becomes unworkable with multiple responses with different protocols, and there is already a preference factor built into NAPTR records that should be accessible from the dialplan when an EnumLookup is returned. Anyone want to take a swing at it? Otmar? :-) JT ___ 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] Calls with incoming distinctive ring
Hello List I have searched the lists, the wiki and the handbook and see how to use distinctive ring inside however I can't find incoming. I have 1 x100p and 2 phone numbers, My Voice calls are normal ring, my Fax are short short long. How do I tell * to route the call to an extension based on the ring candance? Its been in since the .7 release see the dring section in zapata.conf configs/zapata.conf.sample in the srcs ___ 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] Power Over Ethernet for *any* ethernet switch (or hub); product idea
-Original Message- From: Ken Alker [mailto:[EMAIL PROTECTED] Sent: Tuesday, January 20, 2004 2:59 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Power Over Ethernet for *any* ethernet switch (or hub); product idea Does something like this already exist for cheap? If so, is it any good? If so, does it need more features? If not, would you buy something like this? If so, what features have I missed? If so, what is it worth? Daydreaming, as usual. Ken Ken, 3Com makes a 24-port midspan box that sells for around $800. Kevin This electronic message transmission, including attachments, is for the exclusive use of the individuals to which this e-mail is addressed and is to be reviewed and used exclusively for authorized company purposes. This transmission may contain proprietary, confidential or privileged information. If you are not the intended recipient of this transmission, you are hereby notified that any use, copying, disclosure, dissemination, distribution or taking of any action in reliance upon the contents of this transmission is strictly prohibited. If you believe you may have received this electronic message in error, please notify the sender immediately by return email and delete or destroy the original message and/or any copy of it from your computer system and/or your files. Thank you. ___ 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] Compiling problems with SuSE
Did anyone try compiling with optimizations off? I seemed to noticed that the default flag was an O9 or something. Try with -O1 or with -g ans see if it makes any difference. Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dustin Knuttgen Sent: Tuesday, January 20, 2004 9:53 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Compiling problems with SuSE -Original Message- From: Uwe Klein [mailto:[EMAIL PROTECTED] Sent: Monday, January 19, 2004 9:14 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Compiling problems with SuSE From: Dustin Knuttgen on Sunday, January 18, 2004 11:47 PM We tried to use SuSE initially and had no luck compiling zaptel on either 8.2 or 9.0. We even had Digium take a look. After working on it for days we finally switched to Red Hat 9. Is there anyone who succeeded in compiling Asterisk with SuSE 8.2 or 9.0? HI Dustin, what kind of error did you get? something like this: pbx.c:581: warning: comparison between signed and unsigned pbx.c: In function `pbx_substitute_variables_temp': pbx.c:765: warning: comparison between signed and unsigned pbx.c:812: warning: comparison between signed and unsigned pbx.c: In function `pbx_builtin_hangup': pbx.c:4017: internal compiler error: Segmentation fault ?? I had problems with SuSE 8.2 and Asterisk from cvs dated ~12July2003 I got it fixed by adding 128MB of memory to the 32MB on this P200 machine. with 300MB of swap it should not have made a difference ( except taking forever ) but it did. G! UK -- Uwe Klein [mailto:[EMAIL PROTECTED] KLEIN MESSGERAETE Habertwedt 1 D-24376 Groedersby b. Kappeln, GERMANY phone: +49 4642 920 123 FAX: +49 4642 920 125 ___ 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 Uwe, I had a problem at the end when it does the depmod -a. We got an error with around ten modules. The only thing I could find related to the errors was something about PPP in the kernel or in the Makefile. Neither of which made any difference. Dustin ___ 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 -- Notice: This e-mail message, together with any attachments, contains information of Merck Co., Inc. (One Merck Drive, Whitehouse Station, New Jersey, USA 08889), and/or its affiliates (which may be known outside the United States as Merck Frosst, Merck Sharp Dohme or MSD and in Japan, as Banyu) that may be confidential, proprietary copyrighted and/or legally privileged. It is intended solely for the use of the individual or entity named on this message. If you are not the intended recipient, and have received this message in error, please notify us immediately by reply e-mail and then delete it from your system. -- ___ 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] Re-Invite between SIP phones
I suspect you are using a Dial() statement that has something like T or t on it, which will force the media path through Asterisk so that Asterisk can listen for # keypresses. Please include the full context of the dialing routine so it can be examined. Trim down a test to the absolute simplest form of a Dial and try to see if reinvite works. JT At 6:30 AM -0800 1/20/04, Al wrote: I'm trying to place calls between Cisco ATAs and XLite clients. Calls go through perfectly. Both sides of the call negotiate the same CODEC (G711a). I read that older Cisco ATA 186 firmwares don't support reinvites but when capturing traffic there is no Asterisk attempt to send the reinvite message. Al --- Low, Adam [EMAIL PROTECTED] wrote: I'd suggest placing a packet sniffer (tcpdump, etherreal) and see whats happening because it works great for me and always has but I guess it also requires support on the end-points and possibly (assuming non-cisco enviro) there maybe an option that needs to be configured on your phones/gateways. Please provide more information on your setup ... -Original Message- From: Al [mailto:[EMAIL PROTECTED] Sent: Tuesday, January 20, 2004 2:52 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Re-Invite between SIP phones Already did that, but it's not working. Al --- Low, Adam [EMAIL PROTECTED] wrote: canreinvite=yes within sip.conf entities ... -Original Message- From: Al [mailto:[EMAIL PROTECTED] Sent: Tuesday, January 20, 2004 2:06 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Re-Invite between SIP phones Anybody knows what do I need to tell Asterisk to issue a re-INVITE between two SIP phone to avoid having the media going through the server? Tks, Al [People- TRIM YOUR POSTS - there was like 6k worth of crap down here] ___ 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] New sounds also now in CVS
At 8:43 AM + 1/20/04, Miguel A Paraz wrote: On Tue, Jan 20, 2004 at 12:25:46AM -0800, Ken Alker wrote: What other broad topics for words exist right now besides those that are PBX specific and weather-related? I'd like prepaid calling phrases. PIN's, card numbers, account numbers, balance... Insufficient data. Why don't you make a list of EXACTLY what phrases you want to see, and maybe someone will grant you your wish. JT ___ 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] Power Over Ethernet for *any* ethernet switch (or hub); product idea
On Tue, 2004-01-20 at 08:02, Matteo Brancaleoni wrote: Hi. The POEI simply connects the four ethernet signals on each of its inputs (pins 1,2,3,6 on each) to the same pins on its corresponding outputs. Additionally, it supplies -48VDC (maybe selectable if there are other voltage needs) on the appropriate pins (also maybe selectable if different vendors use different wiring conventions for POE) of its outputs. and probably you're going to fry something on your lan. POE isn't simple power on the right pins, but is a sort of protocol. Really, on POE enabled devices (or injectors) you won't measure the DC with a tester, simply because POE on port X is enabled after a request by the device on that port. this is for mantaining compatibity with non POE devices. so you will need also something that detects the power request on each port and enables it. How does a non powered device request power? -- Steven Critchfield [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] Re-Invite between SIP phones
What I would like to understand is in what situations reINVITEs are issued. Anyway, I got the following messages when trying to apply your patch. patching file chan_sip.c Hunk #1 succeeded at 160 with fuzz 1. Hunk #2 succeeded at 365 with fuzz 1. Hunk #4 FAILED at 2253. Hunk #5 FAILED at 2291. Hunk #6 FAILED at 5019. Hunk #7 FAILED at 5168. Hunk #8 succeeded at 5911 with fuzz 1. Hunk #9 FAILED at 6245. Hunk #10 FAILED at 6397. patch unexpectedly ends in middle of line Hunk #11 FAILED at 6683. 7 out of 11 hunks FAILED -- saving rejects to file chan_sip.c.rej Al --- Kannaiyan Natesan [EMAIL PROTECTED] wrote: Hi, I think canreinvite=yes won't work in most of the situations. I have implemented Redirect SIP 300 Message to redirect to the SIP address you speficy in the sip.conf. Where you can have , register = username:[EMAIL PROTECTED]/extension [extension] redirect=yes redirecturi=sip:[EMAIL PROTECTED] redirecturi=sip:[EMAIL PROTECTED] ... will make to redirect to all the URI's yu specify in the sip.conf. I'm also working on this so that it can get the redirections from the database rather than reloading asterisk all the time when you modify the redirection uri. You can check through that. http://bugs.digium.com/bug_view_page.php?bug_id=879 Message transmission is alright, but for some reason it is not working. Can you test with yours and let me know where is the problem, I will modify the code once you get the clue where is the problem on it. If successfully please send me the sip debug message and I will just make sure it works for all. Kannaiyan - Original Message - From: Low, Adam [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, January 20, 2004 2:00 PM Subject: RE: [Asterisk-Users] Re-Invite between SIP phones I'd suggest placing a packet sniffer (tcpdump, etherreal) and see whats happening because it works great for me and always has but I guess it also requires support on the end-points and possibly (assuming non-cisco enviro) there maybe an option that needs to be configured on your phones/gateways. Please provide more information on your setup ... -Original Message- From: Al [mailto:[EMAIL PROTECTED] Sent: Tuesday, January 20, 2004 2:52 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Re-Invite between SIP phones Already did that, but it's not working. Al --- Low, Adam [EMAIL PROTECTED] wrote: canreinvite=yes within sip.conf entities ... -Original Message- From: Al [mailto:[EMAIL PROTECTED] Sent: Tuesday, January 20, 2004 2:06 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Re-Invite between SIP phones Anybody knows what do I need to tell Asterisk to issue a re-INVITE between two SIP phone to avoid having the media going through the server? Tks, Al __ Do you Yahoo!? Yahoo! Hotjobs: Enter the Signing Bonus Sweepstakes http://hotjobs.sweepstakes.yahoo.com/signingbonus ___ 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 * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise 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 ___ 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 __ Do you Yahoo!? Yahoo! Hotjobs: Enter the Signing Bonus Sweepstakes http://hotjobs.sweepstakes.yahoo.com/signingbonus ___ 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 * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise 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
Re: [Asterisk-Users] Power Over Ethernet for *any* ethernet switch (or hub); product idea
On Tue, 2004-01-20 at 02:59, Ken Alker wrote: Based on several threads I've read on this list, I assume that it would be handy to supply POE (power over ethernet) in an environment without having to purchase POE switches (assumed expensive) and abandon one's existing (familiar/custom/not-yet-expensed/etc.) switches/hubs. Assume I have a non-POE switch with 24 RJ-45 (ethernet) ports. I design a 1U box that can be mounted just above/below the non-POE switch, call it a POEI (POE inserter). This box has 48 RJ-45 ports, 24 inputs and 24 outputs. The end user removes all the ethernet cables connected to the existing switch and moves them to the outputs of the POEI. Next, the end user takes six-inch long ethernet cables and connects each (now vacant) port of the existing switch to the inputs of the POEI. The POEI simply connects the four ethernet signals on each of its inputs (pins 1,2,3,6 on each) to the same pins on its corresponding outputs. Additionally, it supplies -48VDC (maybe selectable if there are other voltage needs) on the appropriate pins (also maybe selectable if different vendors use different wiring conventions for POE) of its outputs. This could be an inexpensive way to provide POE without having to replace all of one's switches. Additionally, this could be a nifty business opportunity. Are POE switches expensive enough to warrant manufacturing above? If not, is there a case for not having to swap out all of ones existing switches? Does something like this already exist for cheap? If so, is it any good? If so, does it need more features? If not, would you buy something like this? If so, what features have I missed? If so, what is it worth? Your main problem is going to be in metering. I think the PoE spec is some smallish ma rating. If you use some COTS power supply capable of providing power to all 24 ports, your talking about some pretty hefty power, and unless you wish to put some form of circuitry to act as a limiter per port, your could end up with some nasty problems. Also I believe the spec states -48vdc. IT isn't difficult for a small power regulator on the device side to make that what it needs inside after the voltage drop for distance. -- Steven Critchfield [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] R2 support
Ok, it's old and clunky, but in some countries like Brazil, Argentina and China is the only alternative. Only alternative??? Why is the only alternative? All mayor carriers in Argentina and Brasil have PRI signalling, at the same price. ___ 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] Power Over Ethernet for *any* ethernet switch (or hub); product idea
On Tuesday 20 January 2004 10:30 am, Kevin Ragsdale wrote: 3Com makes a 24-port midspan box that sells for around $800. Kevin This electronic message transmission, including attachments, is for the exclusive use of the individuals to which this e-mail is addressed and is to be reviewed and used exclusively for authorized company purposes. This transmission may contain proprietary, confidential or privileged information. If you are not the intended recipient of this transmission, you are hereby notified that any use, copying, disclosure, dissemination, distribution or taking of any action in reliance upon the contents of this transmission is strictly prohibited. If you believe you may have received this electronic message in error, please notify the sender immediately by return email and delete or destroy the original message and/or any copy of it from your computer system and/or your files. Thank you. http://www.goldmark.org/jeff/stupid-disclaimers/ -- Art is anything you can get away with. -- Marshall McLuhan. ___ 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] ADSI phone vs. IP phone (and proper implementation thereof)
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ray Burkholder Sent: Monday, January 19, 2004 7:38 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] ADSI phone vs. IP phone (and proper implementation thereof) [...] I'm wondering if what you say is actually true. According to recent media releases, Cisco has shipped over 2 million of their IP phones. They must be doing something right. [...] Yes, they are marketing well, and the phones work just fine. But what does the number of units shipped have to do with anything? I've got a dump truck load of 1721/VPN-K9s with ADSL cards that STILL have an open bug after almost six months (which causes blocking on the ATM port, rendering the router unable to pass traffic). Does that mean they are perfect? No. I'd go as far as saying that the 7960s are better than that, as they work very well. Until you try to use the built in switch and hit the right conditions. [...] Voip quality is not necessarily about bandwidth (because it works on T1 data lines as well as GB ports), but about instantaneous bottlenecks in the network. These instantaneous and random bottlenecks can occur in the cad environment mentioned. But with appropriate COS (layer 2) and TOS (layer 3) settings in the phones, switches, and routers, these bottlenecks become non- issues. [...] That's VoIP 101. The real issue is that the phones crash/reboot/degrade under high pps on the switch. Probably because of all of that processing for VLANS and switching taking place on the same processor as the phone (just a guess, I have no idea of the internal design). Go get yourself a nachi-style worm, or other high-pps type app and put it on a reasonable well-powered machine on a 7960. Crank up the packets and try to make phone calls. Then we'll talk again. Daryl G. Jurbala BMPC Network Operations Tel (NY): +1 917 477 0468 x235 Tel (MI): +1 616 608 0004 x235 Tel (UK): +44 208 792 6813 x235 Fax: +1 508 526 8500 INOC-DBA: 26412*DGJ PGP Key: http://www.introspect.net/pgp ___ 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] ADSI phone vs. IP phone
On Mon, 19 Jan 2004, Ted Cabeen waxed: Andrew Kohlsmith [EMAIL PROTECTED] writes: Why wouldn't you just use your existing Ethernet infrastructure putting the IP phones inline between the wall jack and the PC? There are a number of IP phones that have builtin switch/hub that allows the PC to daisy chain off the IP phone. To quote myself: True, but I don't have to retool my office and install POE switches to use ADSI phones, either. No, I will not put a hub/switch at every desk and then use wall-warts for every phone to get around retooling the office. :-) I'm not going to bastardize my network by placing the equivalent of a 3-port switch or hub at every desk to have the phone system compete with our heavy network users (CAD mostly), and I will fight tooth and nail against having to put a goddamned wall-wart at every station just to power the damned IP phones. :-) Do ADSI phones need wall-warts, or can they drive themselves from the line power? You can get dial tone on ADSI w/o a wall-wart, just like a regular analog phone. But you need a wall-wart to give you power for the screen and ADSI functionality, at least on the Nortel Vista 350. Since there's no Ethernet, I don't think it would be practical to do POE. --Chris -- Chris Maj cmaj_hat_freedomcorpse_hot_info Pronunciation Guide: Maj == May Fingerprint: 43D6 799C F6CF F920 6623 DC85 C8A3 CFFE F0DE C146 ___ 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] ADSI phone vs. IP phone
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of PJ Sent: Tuesday, January 20, 2004 5:09 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] ADSI phone vs. IP phone On Mon, 19 Jan 2004, David Gomillion wrote: Andrew wrote: First, what's wrong with PoE? Is it any worse than installing tons of channel banks? Can anybody recommend a good PoE product? I am interested in getting that implemented. You need to be more specificPoE isn't all standard. As is par for the Course, Cisco has their own. So If you're talking about 79xx's, I can definitely recommend any of the PoE blased for the Cat 4500 and 6500 series. Just make sure you have enough wattage coming form your power supplies (I had to go to 220v on one after loading it up with PoE blades). For smaller wiring closets, the Cat 3524-PWR-XL works great. And if you also have a Cisco wireless infrastructure (AiroNet 350 and newer) you can power those with the same hardware. Daryl G. Jurbala BMPC Network Operations Tel (NY): +1 917 477 0468 x235 Tel (MI): +1 616 608 0004 x235 Tel (UK): +44 208 792 6813 x235 Fax: +1 508 526 8500 INOC-DBA: 26412*DGJ PGP Key: http://www.introspect.net/pgp ___ 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] [ot] Grandstream hardware
Eric Wieling wrote: How does Grandstream become patent indemnified for their hardware? I would assume they did not pay for a license for G723,1 and G729 directly to the patent holding company. Maybe they did. I always assumed the indemnification came with a DSP that implemented the codec. Most people buy the codecs as software packages from one of a few companies that specialise in writing major DSP modules. The royalty those companies charge for the software usually includes the patent fees, which they pass on to the patent holders. If you are lucky, they will indemnify the equipment maker that they have paid all relevant charges. :-) Regards, 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
RE: [Asterisk-Users] Power Over Ethernet for *any* ethernet switch (or hub); product idea
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ken Alker Sent: Tuesday, January 20, 2004 3:59 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Power Over Ethernet for *any* ethernet switch (or hub); product idea [...] Assume I have a non-POE switch with 24 RJ-45 (ethernet) ports. I design a 1U box that can be mounted just above/below the non-POE switch, call it a POEI (POE inserter). This box has 48 RJ-45 ports, 24 [...] Are POE switches expensive enough to warrant manufacturing above? If not, is there a case for not having to swap out all of ones existing switches? [...] Depends on what expensive means, and whether your switces are due for replacemtn or not. And what you intended to replace them with not counting PoE. The difference between Catalyst 2950-XL-24s and 3524-PRW-XL's is about $300. The difference on a large Catalyst switch is about $5-10/port if I recall correctly from my last deployment. Does something like this already exist for cheap? Yes. Several. If so, is it any good? Yes. Many work just fine. If so, does it need more features? To do what? It's called a mid-span power injector. The ones I've seen do that and nothing else. I'd say they are living up to their task. If not, would you buy something like this? If so, what features have I missed? If so, what is it worth? Google the rest of your answers. You're about 6 years too late to catch the first run of this train. Daryl G. Jurbala BMPC Network Operations Tel (NY): +1 917 477 0468 x235 Tel (MI): +1 616 608 0004 x235 Tel (UK): +44 208 792 6813 x235 Fax: +1 508 526 8500 INOC-DBA: 26412*DGJ PGP Key: http://www.introspect.net/pgp ___ 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] ultra-cheap asterisk box
On Tuesday 20 January 2004 03:22 am, Ken Alker wrote: Radio Shack has a really neat A/C power meter that plugs into the wall. You then plug the A/C powered appliance you want to test into the unit. The unit reports instantaneous KW, KVA, and power consumption over time. It claims 15A max, but I've run it much higher. This is a great tool and a really fun gadget as well! It is one of those things you just don't want to spend the money on for a single test, but then when you have it you find uses for it constantly. I went around measuring everything in the house and at work after I got it. I think it was about $50, but on sale it was a good 30% off! Does it allow for the power factor? Brand have been producing good power meters for several years now. Prices are not as low as the above but some models have remote capability. http://www.brandelectronics.com/ Regards...Martin -- A straw vote only shows which way the hot air blows. -- O'Henry ___ 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] Enter Pin followed by Pound key
Im trying to create a custom application via the AGI. I want to authenticate the users that dial in with a userid and pin. However, the number of digits in the PIN and userid are variable, and therefore I need to allow the user to press enter by hitting the pound key. How would I accomplish this in the AGI? stream_file doesnt seem to work, since it only allows one digit to be pressed. get_data seems to only allow a fixed number of digits to be entered. Thanks Gary Franczyk ___ 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] PSTN Gateway
Hello,
I am looking for information on setting up digium FXO card for use as a
PSTN Gateway (H323-PSTN) to work with GNUGk.
I am basically looking for the setup and it would be great if anyone can
share his experiences with the same. Also, if there are any limitations in
going for such a setup and problems that may arise/things that I should keep
in consideration.
Thanks Regards,
Deepak
- Original Message -
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To: [EMAIL PROTECTED]
Sent: Tuesday, January 20, 2004 12:08 PM
Subject: Asterisk-Users digest, Vol 1 #2557 - 10 msgs
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than Re: Contents of Asterisk-Users digest...
Today's Topics:
1. FW: Memory problem (T. Chan)
2. X101P CallOut Big Problem. (Carlos Arnt)
3. RE: RE: Latest version of asterisk (Aram Ter-Martirosyan)
4. Re: SIP: Register that isn't a register? (Ing. Angel Gomez Garcia)
5. RE: FW: Memory problem (Adam Goryachev)
6. Call token is ip$localhost (Asan M.)
7. Re: CVS Changes (NAT-SIP) (Brian West)
8. Re: PLAYBACK multiple files (Marcin Kuzmicki)
9. Re: user password and call waiting (Brian West)
10. echo cancellation (dkwok)
--__--__--
Message: 1
From: T. Chan [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Date: Mon, 19 Jan 2004 23:20:27 -0500
Subject: [Asterisk-Users] FW: Memory problem
Reply-To: [EMAIL PROTECTED]
Dear all,
I have had an experience which I would run by all of you to see if this is
normal.
I am running a few asterisk servers with 512M RAM memory, and as I have
mentioned in previous notes, I have experienced frequent crashes when
faced
with more than 15-20 simultaneous calls. I have tried to find out if it
could be due to (a) Xeon chip running HT, (b) old Kernel version 2.4.18-3,
(c) old redhat linux version 7.3, (d) H323 library pwlib and openh323
versions which are 1.5.2 and 1.12.2 respectively among many other
parameters. So far, unfortunately, the matter has not been resolved.
However, I have noticed that the memory usage on each server has built up
with time after the server being rebooted. I have complained about using
close to 500M even when there were very few calls on the server but nobody
seemed to be able to let me know if they were running at high memory
usages
except for Jesse who was telling me that his memory usages have always
been
low. Very recently, I noticed that after I rebooted the servers, the
memory
usage would start at about 80 M and even after started the Asterisk
threads,
I was running at about 100 M and even when there were calls, I was running
at about 100M-150M, but then after hours it would start to build up to
200M
and then 250M and thenfinally close to 500M even after I stopped the
Asterisk threads, almost like there is a memory leak somewhere.
I wonder if that is normal, if someone can please tell me, or if not
normal,
what could be the cause to it and how should this be rectified.
Thanks alot
Tom
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Message: 2
From: Carlos Arnt [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Date: Tue, 20 Jan 2004 02:39:09 -0200
Subject: [Asterisk-Users] X101P CallOut Big Problem.
Reply-To: [EMAIL PROTECTED]
htmlheadmeta name=3DGenerator content=3DPocoMail 3 HTML/CSS=
Generator/
style type=3Dtext/css!--
LI{display:list-item;margin:0.00in;}
p{display:block;margin:0.00in;}
body{}
--/style
/headBODY pSPAN style=3Dfont-size:10pt;Hi all,/SPAN/p
pnbsp;/p
pSPAN style=3Dfont-size:10pt;I just now receive the FXO X101P Card
but=
can't at any way make then call out./SPAN/p
pSPAN style=3Dfont-size:10pt;I can hear the signal, even call but
always=
receive from my local operator error that or the number don't exist or
need=
more numbers./SPAN/p
pnbsp;/p
pSPAN style=3Dfont-size:10pt;I play alot with txgain and rxgain,
but=
none help me out./SPAN/p
pSPAN style=3Dfont-size:10pt;Being honest i try alot 5 hours
and=
none !!!/SPAN/p
pnbsp;/p
pSPAN style=3Dfont-size:10pt;I'm using asterisk in his sample=
configs./SPAN/p
pSPAN style=3Dfont-size:10pt;I mean i call out using 1234=
etc../SPAN/p
pSPAN style=3Dfont-size:10pt;Zapata.conf is Ok/SPAN/p
pSPAN style=3Dfont-size:10pt;Zaptel.conf is ok/SPAN/p
pSPAN style=3Dfont-size:10pt;(I follow the Digium faqs, then for a
good=
person
RE: [Asterisk-Users] Power Over Ethernet for *any* ethernet switch (or hub); product idea
-Original Message- From: Martin [mailto:[EMAIL PROTECTED] Sent: Tuesday, January 20, 2004 10:23 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Power Over Ethernet for *any* ethernet switch (or hub); product idea snip http://www.goldmark.org/jeff/stupid-disclaimers/ -- Art is anything you can get away with. -- Marshall McLuhan. Martin, We have rules in place that remove it from emails to mailing lists, but I fat-fingered the digium address. Should be fixed now. Apologies, Kevin ___ 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] Re-Invite between SIP phones
basically for (re)negotiation on session parameters like: media codecs, media IP and PORT In common it is useful to put a line on hold by setting the media IP to 0.0.0.0 or for a soft redirection of the media stream to another IP and/or PORT. On the second hand there is a feature called "timer" to check for aliveness of a session. Al wrote: What I would like to understand is in what situations reINVITEs are issued. Anyway, I got the following messages when trying to apply your patch. patching file chan_sip.c Hunk #1 succeeded at 160 with fuzz 1. Hunk #2 succeeded at 365 with fuzz 1. Hunk #4 FAILED at 2253. Hunk #5 FAILED at 2291. Hunk #6 FAILED at 5019. Hunk #7 FAILED at 5168. Hunk #8 succeeded at 5911 with fuzz 1. Hunk #9 FAILED at 6245. Hunk #10 FAILED at 6397. patch unexpectedly ends in middle of line Hunk #11 FAILED at 6683. 7 out of 11 hunks FAILED -- saving rejects to file chan_sip.c.rej Al --- Kannaiyan Natesan [EMAIL PROTECTED] wrote: Hi, I think canreinvite=yes won't work in most of the situations. I have implemented Redirect SIP 300 Message to redirect to the SIP address you speficy in the sip.conf. Where you can have , register = username:[EMAIL PROTECTED]/extension [extension] redirect=yes redirecturi=sip:[EMAIL PROTECTED] redirecturi=sip:[EMAIL PROTECTED] ... will make to redirect to all the URI's yu specify in the sip.conf. I'm also working on this so that it can get the redirections from the database rather than reloading asterisk all the time when you modify the redirection uri. You can check through that. http://bugs.digium.com/bug_view_page.php?bug_id=879 Message transmission is alright, but for some reason it is not working. Can you test with yours and let me know where is the problem, I will modify the code once you get the clue where is the problem on it. If successfully please send me the sip debug message and I will just make sure it works for all. Kannaiyan - Original Message - From: "Low, Adam" [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, January 20, 2004 2:00 PM Subject: RE: [Asterisk-Users] Re-Invite between SIP phones I'd suggest placing a packet sniffer (tcpdump, etherreal) and see whats happening because it works great for me and always has but I guess it also requires support on the end-points and possibly (assuming non-cisco enviro) there maybe an option that needs to be configured on your phones/gateways. Please provide more information on your setup ... -Original Message- From: Al [mailto:[EMAIL PROTECTED]] Sent: Tuesday, January 20, 2004 2:52 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Re-Invite between SIP phones Already did that, but it's not working. Al --- "Low, Adam" [EMAIL PROTECTED] wrote: canreinvite=yes within sip.conf entities ... -Original Message- From: Al [mailto:[EMAIL PROTECTED]] Sent: Tuesday, January 20, 2004 2:06 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Re-Invite between SIP phones Anybody knows what do I need to tell Asterisk to issue a re-INVITE between two SIP phone to avoid having the media going through the server? Tks, Al __ Do you Yahoo!? Yahoo! Hotjobs: Enter the "Signing Bonus" Sweepstakes http://hotjobs.sweepstakes.yahoo.com/signingbonus ___ 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 * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise 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 ___ 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 __ Do you Yahoo!? Yahoo! Hotjobs: Enter the
RE: [Asterisk-Users] R2 support
yes but PRI is not a trunk, R2 can be used as a trunk. Luciano -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] nombre de CW_ASN - Gus Enviado el: Martes 20 de Enero del 2004 13:08 Para: [EMAIL PROTECTED] Asunto: Re: [Asterisk-Users] R2 support Ok, it's old and clunky, but in some countries like Brazil, Argentina and China is the only alternative. Only alternative??? Why is the only alternative? All mayor carriers in Argentina and Brasil have PRI signalling, at the same price. ___ 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 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] CAPI: Early-B3 working with AVM-B1?
Hi,
I tested the capi_chan with latest cvs of * and I have problems with
Early-B3. The following dialstring works for me (without Early B3):
exten = _0X.,1,Dial(CAPI/@22715291:${EXTEN:1}|30)
But if I add the 'b' for using Early-B3
exten = _0X.,1,Dial(CAPI/@22715291:b${EXTEN:1}|30)
nothing changes (no dialtone). If in this example the called party
discards the call, there is no signalling to my SIP-Phones. In this case
capi debug tells a lot of:
-- CONNECT_B3_ACTIVE_IND ID=001 #0xb2f4 LEN=0013
Controller/PLCI/NCCI= 0x10101
NCPI= default
sent CONNECT_B3_ACTIVE_RESP (NCCI=0x10101)
sent DATA_B3_RESP (NCCI=0x10101)
sent DATA_B3_RESP (NCCI=0x10101)
sent DATA_B3_RESP (NCCI=0x10101)
sent DATA_B3_RESP (NCCI=0x10101)
sent DATA_B3_RESP (NCCI=0x10101)
sent DATA_B3_RESP (NCCI=0x10101)
sent DATA_B3_RESP (NCCI=0x10101)
sent DATA_B3_RESP (NCCI=0x10101)
sent DATA_B3_RESP (NCCI=0x10101)
sent DATA_B3_RESP (NCCI=0x10101)
sent DATA_B3_RESP (NCCI=0x10101)
sent DATA_B3_RESP (NCCI=0x10101)
... until I stop the call from SIP phone (the originating site)
-- CAPI Hangingup
activehangingup
sent DISCONNECT_B3_REQ NCCI=0x10101
-- DISCONNECT_B3_CONF ID=001 #0x001a LEN=0014
Controller/PLCI/NCCI= 0x10101
Info= 0x0
== DISCONNECT_B3_IND NCCI=0x10101
sent DISCONNECT_B3_RESP NCCI=0x10101
sent DISCONNECT_REQ PLCI=0x101
-- DISCONNECT_CONF ID=001 #0x001b LEN=0014
Controller/PLCI/NCCI= 0x101
Info= 0x0
== DISCONNECT_IND PLCI=0x101 REASON=0x3495
sent DISCONNECT_RESP PLCI=0x101
-- removed pipe for PLCI = 0x101
---
Hardware is a AVM-B1 (active BRI card)
What am I doing wrong, or where can I start debugging?
Thanks,
Karsten
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RE: [Asterisk-Users] Power Over Ethernet for *any* ethernet switch(or hub); product idea
PoE, or 802.3af, uses a device detection routine to determine if the connected device needs power. The process, in greatly simplified terms, is as follows: 1. Detect link state 2. Send a pulse of a known frequency and intensity over the TX/RX pairs 3. Listen for reflection. 3a. No reflection- provide power 3b. Reflection- no power Devices that comply with 802.3af have filters designed into the TX/RX paths to block the detection pulses, thereby identifing themselves as able to use PoE. The detection process is passive on the device, since if it has no power it cannot 'signal' that it needs power. The process is repeated several times a second to ensure that a PoE is not unplugged and a non-PoE is plugged into it's place and damaged. Issues with midspans devices: The 24 port models are usually 12 port in reality. 12 in and 12 out. Sure there are 24 ports, but you are only going to power 12 devices. So in a larger environment they quickly get expensive. To make the whole situation more interesting the Cisco phones support not only 802.3af, but Cisco's own spin on inline power, which is similar in design to 802.3af. Dan -Original Message- From: Steven Critchfield [mailto:[EMAIL PROTECTED] Sent: Tuesday, January 20, 2004 7:55 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Power Over Ethernet for *any* ethernet switch(or hub); product idea On Tue, 2004-01-20 at 08:02, Matteo Brancaleoni wrote: Hi. The POEI simply connects the four ethernet signals on each of its inputs (pins 1,2,3,6 on each) to the same pins on its corresponding outputs. Additionally, it supplies -48VDC (maybe selectable if there are other voltage needs) on the appropriate pins (also maybe selectable if different vendors use different wiring conventions for POE) of its outputs. and probably you're going to fry something on your lan. POE isn't simple power on the right pins, but is a sort of protocol. Really, on POE enabled devices (or injectors) you won't measure the DC with a tester, simply because POE on port X is enabled after a request by the device on that port. this is for mantaining compatibity with non POE devices. so you will need also something that detects the power request on each port and enables it. How does a non powered device request power? -- Steven Critchfield [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 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] Power Over Ethernet for *any* ethernet switc h (or hub); product idea
That'd be the 3CNJP24SE, we have one that powers 3COM NJ-200's. Works well. -Original Message- From: Martin [mailto:[EMAIL PROTECTED] Sent: Tuesday, January 20, 2004 9:23 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Power Over Ethernet for *any* ethernet switch (or hub); product idea On Tuesday 20 January 2004 10:30 am, Kevin Ragsdale wrote: 3Com makes a 24-port midspan box that sells for around $800. Kevin This electronic message transmission, including attachments, is for the exclusive use of the individuals to which this e-mail is addressed and is to be reviewed and used exclusively for authorized company purposes. This transmission may contain proprietary, confidential or privileged information. If you are not the intended recipient of this transmission, you are hereby notified that any use, copying, disclosure, dissemination, distribution or taking of any action in reliance upon the contents of this transmission is strictly prohibited. If you believe you may have received this electronic message in error, please notify the sender immediately by return email and delete or destroy the original message and/or any copy of it from your computer system and/or your files. Thank you. http://www.goldmark.org/jeff/stupid-disclaimers/ -- Art is anything you can get away with. -- Marshall McLuhan. ___ 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 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] wav49 voicemail problem with Windows Media Player
[Sorry if this gets posted twice -- I sent it with the wrong account and it's stuck in moderator review...] I downloaded the files from the bug tracker and had a look at them. The original msg.WAV is slightly malformed: it's chunk tags are too big. A lot of audio programs ignore this because it's easy to get wrong when writing a WAV file. But Media Player cuts no slack. See my post at bugs.digium.com for more info. I'll snoop around for the bug in the source... as soon as I manage to get * to record sound at all. On Fri, 16 Jan 2004 10:19:19 -0500 Jim Flagg [EMAIL PROTECTED] wrote: I have done some more investigating and posted this in Bug Tracker I have found that the Microsoft Sound Recorder will play the original posted wave file msg.WAV without errors. I opened this file and then re-saved it inside of Sound Recorder with the same GSM 6.10 (wav49) format. The resulting file (msga.WAV) is slightly different than the original. The msga.WAV file plays without error on Windows Media Player. Maybe this will give someone a hint as to what the problem is. ___ 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] MeetMe questions
I'm looking into deploying * for an internal conference call server (using MeetMe) and had a couple of quick questions for those of you who have used it. I checked the Wiki but there weren't a lot of details for MeetMe. - Can you limit the size of a conference room, ie max 8 people, etc. - Is there a list somewhere (besides the source ;) that has all the commands availible to people in the conferences? Specifically can you do a mute all new callers type action (when people are really just calling up to listen. - Passwords/Pins for the conference rooms? Thanks all, Chris Robertson Network Engineer Instill Corp. ___ 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] DTMF A-D
--On Monday, January 19, 2004 11:01 AM -0500 Andrew Kohlsmith [EMAIL PROTECTED] wrote: SNIP'd from the ADSI phone vs. IP phone thread I'm looking at ADSI phones simply because I don't have to re-tool my entire building; I can use the existing phone network and (I think) get all the functionality I need with the (far) cheaper ADSI phones. My basic ADSI functionality is - (assisted/consultative and blind) transfers - voicemail integration (next/prev/forward, MWI, etc.) - caller ID display - conference - hold/park/pickup - paging - handsfree - DND - global and per-extension speed dial - muting of DTMF A-D from the far-end I've know about DTMF A-D for 20+ years now, but have never heard anyone mention it before, or use it, for that matter (except in old silver boxing in the bad ol' days). Can you elaborate upon how you'd take advantage of DTMF A-D, how you'd produce the tones (are these standard now?), and what exactly you mean by muting from the far-end? ___ 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] Enter Pin followed by Pound key
- Original Message -
From: Gary Franczyk [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, January 20, 2004 10:51 AM
Subject: [Asterisk-Users] Enter Pin followed by Pound key
Im trying to create a custom application via the AGI. I want to
authenticate the users that dial in with a userid and pin. However, the
number of digits in the PIN and userid are variable, and therefore I need
to
allow the user to press enter by hitting the pound key. How would I
accomplish this in the AGI?
stream_file doesnt seem to work, since it only allows one digit to be
pressed.
get_data seems to only allow a fixed number of digits to be entered.
Sorry if I'm speaking out of school, as I have never programmed AGIs, but
from what you described, the stream_file taking one digit at a time should
be sufficient.
string entry
while (keypressed != #)
{
entry += keypressed
}
In this way, you could build up your string of digits.
Don't know how AGI is working specifically, but hopefully this will trigger
some thought or idea.
Thanks
Gary Franczyk
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Re: [Asterisk-Users] ADSI phone vs. IP phone
--On Monday, January 19, 2004 11:01 AM -0500 Andrew Kohlsmith [EMAIL PROTECTED] wrote: SNIP IP phones are nice, I'll give them that... but they are also a pain in the ass if you're upgrading/retrofiting an office, and they also don't play well together -- you're more or less stuck using one brand of POE switch with one brand of IP phone, or you use wall-warts. ADSI phones feel much more phone-like to me, even though IP Phones can do some wild things. Andrew, If I read above correctly, you imply that ADSI phones don't need wall-warts (A/C power transformers that plug into the wall). I'd assume that based on the sizable LCD screen, potential back-lighting, microcontroller(s), etc, that an ADSI phone would have to have a wall-wart, especially if you wanted to use any of its functionality while it is on-hook. I have designed a phone or two in my past (many years ago) and, as I recall, there is almost *no* current available from the telco while a phone is on-hook. You might be able to trickle-charge a very small battery, or run an RCA 1802 processor (microamps), but that's about it. Did I read your statement correctly, or do ADSI phones truly require wall-warts (as do SIP phones)? ___ 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] G.729 Licenses from Digium
According to digium's site, Note: Please do not attempt to use the G.729 code in a SCSI-only system. We are currently working with VoiceAge to correct this issue. (found at http://www.digium.com/index.php?menu=asterisk_g729). Does anyone know what these issues are? Can anyone define SCSI-only system? I know this sounds kinda dumb, but I have a server with SCSI and IDE interfaces, but no IDE drives. Is that SCSI only? Thanks for your help, David Gomillion ___ 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] How to diagnose pops and clicks?
My setup is as follows: Handset - Sipura SPA 2000 - Asterisk - VoicePulse and Handset - Sipura SPA 2000 - Asterisk - Digium X100P - POTS I notice when making VoicePulse calls (but *not* POTS calls through the X100P) that there is significant popping and clicking on the line. This isn't enough to interfere seriously with the call, and the voice quality is otherwise telephone quality. People I'm calling to don't report hearing the pops and clicks on their end. I'm looking for advice as to how to best diagnose this problem. Thanks, Peter Rukavina Charlottetown, PEI www.reinvented.net ___ 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] help - recording both sides of a conversation
Hello Sirs..
I'm setting up a call-recording with my asterisk here and I do follow
program which was post in this mailing list last Jan. 4 (program is also
shown below), and I'm very much thankful for that..
However, I do have some errors, here is my output..Hope that someone could
lighten me up for this..Thank you very much for the help..
Regards
Joel
*CLI -- Starting simple switch on 'Zap/49-1'
-- Executing Answer(Zap/49-1, ) in new stack
-- Executing Macro(Zap/49-1, record-enable) in new stack
-- Executing AGI(Zap/49-1, set-timestamp.agi) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/set-timestamp.agi
-- AGI Script set-timestamp.agi completed, returning 0
-- Executing Dial(Zap/49-1, Zap/51|15) in new stack
-- Called 51
-- Zap/51-1 is ringing
-- Zap/51-1 answered Zap/49-1
-- Attempting native bridge of Zap/49-1 and Zap/51-1
-- Hungup 'Zap/51-1'
== Spawn extension (test3, 2103, 3) exited non-zero on 'Zap/49-1'
-- Executing Macro(Zap/49-1, record-cleanup) in new stack
-- Executing SetVar(Zap/49-1,
MONITORDIR=/var/spool/asterisk/conversations/) in new stack
-- Executing GotoIf(Zap/49-1, = ?6:3) in new stack
-- Goto (macro-record-cleanup,s,3)
Jan 20 13:43:37 WARNING[1256444864]: pbx.c:1173 pbx_extension_helper: No
application 'System(/usr/scripts/mix_monitor_files.pl ${MONITORDIR}
${CALLFILENAME}-in.wav' for extension (macro-record-cleanup, s, 3)
== Spawn extension (macro-record-cleanup, s, 3) exited non-zero on
'Zap/49-1'
in macro 'record-cleanup'
== Spawn extension (test3, h, 1) exited non-zero on 'Zap/49-1'
-- Hungup 'Zap/49-1'
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of zoa
Sent: Tuesday, January 06, 2004 1:05 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] help - recording both sides of a
conversation
You also don't need such a complicated perl script, just muxing them
without cutting them is enough.
(Timing was fixed)
zoa.
At 14:41 4/01/2004 -0600, you wrote:
you nolonger need set-timestamp.agi as we have ${TIMESTAMP} in that format
by default now.
bkw
On Sun, 4 Jan 2004, John Baker wrote:
Iain -
First off, all of this is heavily borrowed from others. For those who
see
their code embedded here, I thank you and give you full credit.
Here's how I do it. It's a bit convoluted, but I didn't want to record
everything. So, if a call comes in and I want to record it, I send it
here:
[ext-surrept]
exten = _57XXX,1,Answer
exten = _57XXX,2,Macro(record-enable)
exten = _57XXX,3,BackGround(for-quality-purposes)
exten = _57XXX,4,BackGround(this-call-may-be)
exten = _57XXX,5,BackGround(recorded)
exten = _57XXX,6,Dial(SIP/${EXTEN:1},120,tm)
exten = _57XXX,7,Macro(rg-inbound,10,tr)
exten = _57XXX,8,Goto(aa-nooneavail,s,1)
By transferring a call to 5 + the extension I'm at, I enable the call
recording, let the caller know he might be recorded and then send the
call
right back to myself.
Here's the Macro:
[macro-record-enable]
exten = s,1,AGI(set-timestamp.agi)
exten =
s,2,SetVar(CALLFILENAME=${timestamp}-${CALLERIDNUM}-${MACRO_EXTEN})
exten = s,3,Monitor(wav,${CALLFILENAME})
It starts the recording and calls set-timestamp.agi
Here's the agi file:
#!/bin/sh
longtime=`date +%Y%m%d-%H%M%S`
echo SET VARIABLE timestamp $longtime
It sets a timestamp, which if you scour the asterisk list, you'll see
that
it is necessary for mixing the in and out audio later.
I have one hangup extension set for my internal phones; it looks like
this:
exten = h,1,Macro(record-cleanup)
And the record-cleanup macro looks like this:
[macro-record-cleanup]
exten = s,1,SetVar(MONITORDIR=/var/spool/asterisk/monitor)
exten = s,2,GotoIf($[${CALLFILENAME} = ${FOO}]?6:3)
exten = s,3,System(/usr/scripts/mix_monitor_files.pl ${MONITORDIR}
${CALLFILENAME}-in.wav ${CALLFILENAME}-out.wav ${CALLFILENAME}.wav)
exten = s,6,NoOp
Don't forget to make the /var/spool/asterisk/monitor directory!
Finally, mix_monitor_files.pl does the mixing job and combines the in
and
out files:
#!/usr/bin/perl
$monitordir = shift;
$infile = shift;
$outfile = shift;
$finishfile = shift;
chdir($monitordir);
$infile_output = `sox $infile -e stat 21`;
$outfile_output = `sox $outfile -e stat 21`;
$infile_output =~ /Samples read:\s+(\d+)/;
$infile_samples = $1;
$outfile_output =~ /Samples read:\s+(\d+)/;
$outfile_samples = $1;
if($outfile_samples $infile_samples)
{
$diff_samples = $outfile_samples - $infile_samples;
system(sox -v 3 $outfile temp${outfile} trim
${diff_samples}s);
system(wmix $infile temp${outfile} $finishfile);
system(rm -f $infile temp${outfile} $outfile);
}
elsif($infile_samples $outfile_samples)
{
$diff_samples = $infile_samples - $outfile_samples;
system(sox -v 3 $infile
[Asterisk-Users] [A-bit-OT] Power Over Ethernet Discovery process
Hi, Since someone asked, here's how POE standard does discovery process for a POE device. of course is a passive detection... but that's why you don't have POE always-on on a POE enabled switch port you can find more info in article area of http://www.poweroverethernet.com and full specs @ http://www.ieee802.org/3/af/index.html You will find a resistance value in the quote below. The value is 19k to 26.5k for PSE detection signature, with a mid value of 22.75K quote The Discovery Process Power Over Ethernet PSEs are responsible for ensuring that conventional Ethernet equipment is not damaged by the unexpected application of 48 Volts. The PSEs must determine that a Power Over Ethernet compliant device is present before the 48V is applied. This is done by the discovery process. A relatively low voltage, current limited, is applied to the CAT-5 cable periodically. A compliant device is required to have a certain DC resistance between its twisted pairs. If the device presents this resistance then power can be applied, but if it does not then power is not applied. The PSE is responsible for monitoring the Powered Device, to check that it is continuing to draw power within certain limits. If it does not (when it is unplugged, for example) then the PSE must remove the power to that cable and return to the discovery stage again. The Powered Device may optionally support a classification mechanism, by which it can signal how much power it will require from the PSE. This allows for better management of what may be a limited power source within the PSE. /quote -- Brancaleoni Matteo [EMAIL PROTECTED] Espia - Emmegi Srl ___ 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] Re-Invite between SIP phones
You are correct. T and t removed. Now reINVITE works. Tks! --- John Todd [EMAIL PROTECTED] wrote: I suspect you are using a Dial() statement that has something like T or t on it, which will force the media path through Asterisk so that Asterisk can listen for # keypresses. Please include the full context of the dialing routine so it can be examined. Trim down a test to the absolute simplest form of a Dial and try to see if reinvite works. JT At 6:30 AM -0800 1/20/04, Al wrote: I'm trying to place calls between Cisco ATAs and XLite clients. Calls go through perfectly. Both sides of the call negotiate the same CODEC (G711a). I read that older Cisco ATA 186 firmwares don't support reinvites but when capturing traffic there is no Asterisk attempt to send the reinvite message. Al --- Low, Adam [EMAIL PROTECTED] wrote: I'd suggest placing a packet sniffer (tcpdump, etherreal) and see whats happening because it works great for me and always has but I guess it also requires support on the end-points and possibly (assuming non-cisco enviro) there maybe an option that needs to be configured on your phones/gateways. Please provide more information on your setup ... -Original Message- From: Al [mailto:[EMAIL PROTECTED] Sent: Tuesday, January 20, 2004 2:52 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Re-Invite between SIP phones Already did that, but it's not working. Al --- Low, Adam [EMAIL PROTECTED] wrote: canreinvite=yes within sip.conf entities ... -Original Message- From: Al [mailto:[EMAIL PROTECTED] Sent: Tuesday, January 20, 2004 2:06 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Re-Invite between SIP phones Anybody knows what do I need to tell Asterisk to issue a re-INVITE between two SIP phone to avoid having the media going through the server? Tks, Al [People- TRIM YOUR POSTS - there was like 6k worth of crap down here] ___ 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 __ Do you Yahoo!? Yahoo! Hotjobs: Enter the Signing Bonus Sweepstakes http://hotjobs.sweepstakes.yahoo.com/signingbonus ___ 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] Remote reload Cisco 7960
I have made the change to my syncinfo.xml file, but still nothing. I have noticed that the phone never looks for that file on the tftp server. Is it possible that the phone is not idle long enough for it to look for the file? Is there a way to check? B. J. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Low, Adam Sent: Tuesday, January 20, 2004 4:03 To: '[EMAIL PROTECTED]' Subject: RE: [Asterisk-Users] Remote reload Cisco 7960 You need a little more to make this script reboot the phone. It basically instructs the phone to check a file called 'syncinfo.xml' at its TFTP URL. This file needs to contain the following line: IMAGE VERSION=* SYNC=2/ The number 2 above is the sync value which must be different (I think higher) than the sync: field defined in your SIPDefault.cnf file. Then the script should do its stuff and reboot the phone. Rgds, Adam -Original Message- From: B. J. Bomar [mailto:[EMAIL PROTECTED] Sent: Monday, January 19, 2004 6:42 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Remote reload Cisco 7960 I've tried to use that script, but the phones seem to ignore it. I am in the process of upgrading to 6.1 on the phones, maybe they will behave like they're supposed to. B. J. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian West Sent: Friday, January 16, 2004 22:27 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Remote reload Cisco 7960 http://www.bkw.org/~brian/cisco/reboot7960.txt or you can us this handy perl script.. NEXT!!! bkw On Fri, 16 Jan 2004, Rich Adamson wrote: Does anyone have a working way of having a Cisco 7960 reload its config remotely. I have tried some of the scripts that I have found on the web, but to no avail. Thanks for the help. telnet to the box and reload it. command line has the ability. 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 ___ 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 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 * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise 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 ___ 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 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] ADSI phone vs. IP phone
Do ADSI phones need wall-warts, or can they drive themselves from the line power? You can get dial tone on ADSI w/o a wall-wart, just like a regular analog phone. But you need a wall-wart to give you power for the screen and ADSI functionality, at least on the Nortel Vista 350. Since there's no Ethernet, I don't think it would be practical to do POE. I thought you were wrong here, as I have Vista 390 at home and I was sure that wasn't the case. Lo and behold one of the biggest reasons for my wanting to go ADSI over IP has been shattered. This is a serious setback for me. :-( Dammit. Blindsided. Regards, Andrew ___ 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] Enter Pin followed by Pound key
Hi! Im trying to create a custom application via the AGI. I want to authenticate the users that dial in with a userid and pin. However, the number of digits in the PIN and userid are variable, and therefore I need to allow the user to press enter by hitting the pound key. How would I accomplish this in the AGI? Did you look at the appliation Digit()? If you must use AGI then EXEC Digit ... might do it for you. stream_file doesnt seem to work, since it only allows one digit to be pressed. get_data seems to only allow a fixed number of digits to be entered. Why not use either of those repetitively and check in your AGI script if # was the digit, then accumlate what you have? Cheers, Philipp ___ 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] Re: Power Over Ethernet for *any* ethernet switch (or hub); product idea
Steven Critchfield wrote: so you will need also something that detects the power request on each port and enables it. How does a non powered device request power? As far as I know, it doesn't. The POE source somehow monitors the line (using impedance, etc) to determine if there is anything connected to the pairs used to supply power. Since netcards and net connected equipment are not supposed to use the power pairs, this should work in most cases. If the termination just leaves them unconnected the impedance is infinity and power will not be applied. Conversely, if the termination are all grounded, impedance is (near) 0 and power will not be applied. For some range of impedances in between, the far end is assumed to require power and power will be applied. Stephen R. Besch ___ 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] Music on Hold - can it be done without mpg123?
I have been having periodic trouble with mpg123. I have tried .59r .59s and perhaps others a while back and still get the 'broken pipe' and zombie mpg123s (although I think I saw something about a fix in the changelog) once and a while. Is it currently possible to configure moh to run directly on the wav files? Now-a-days hard drives are so big, why use compression at all (at least for local files)? John Harragin This e-mail was scanned and found clean by Monroe-Woodbury's Antivirus. ___ 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] Grandstream cfg.txt hacking?
Hi list, I'm trying to figure out the format of the binary data in cfg.txt - so i'm looking for someone with a GS phone/adapter from sipphone.com (bought there so it downloads the config there also). I suppose they use GAPS there and also download the cfg.txt configuration there? Please send tcpdump data of the tftp session - or other udp dump data. I have a tftpd server hacked and ready to serve the configs if we can just get the format of the file. Cheers, Jens Davidsen ___ 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] Agent timeout then Dial() ?
Hello, I have agents / queues working to the extent that agents can login, logout and I can send a caller into the queue and the logged in agent's phones will ring. Maybe I've spent to much time googleing and reading and my eyes are crossing now, but what I am trying to do is this but cannot find any reference to it. 1. Xfer the caller into the Queue... If Noone is logged into the queue, the caller will be directed to a PSTN number instead (or extension, same thing) 2. Xfer the caller into the Queue... Agents are logged in, but the call times out for whatever reason, I would then like to have it go to an extension as in above 3. When say 6PM rolls around and all agents are gone I would like to automagically log them out just incase they forgot to. I will be happy with an answer for 1 and 2 - I can always use a big stick for #3 :) I did find a reference to adding a member local in queues.conf eg: member = local/[EMAIL PROTECTED],10 And have a context in extensions.conf like this [timeout] exten = s,1,Wait,1 exten = s,2,Answer exten = s,3,Playback(transferring_you_offsite) exten = s,4,Dial,IAX2/office/[EMAIL PROTECTED] Even with the metric of '10' to try and give the local member less preference it will give logged in agents like half a ring and then xfer to the timeout context right away. Any help, pointers would be greatly appreciated. Many thanks -bh -- This message was sent using IMP, the Internet Messaging Program. -- This message has been scanned for viruses and dangerous content by the Bugs.Hamel.Net MailScanner, and appears to be clean. ___ 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] DTMF A-D
I've know about DTMF A-D for 20+ years now, but have never heard anyone mention it before, or use it, for that matter (except in old silver boxing in the bad ol' days). Can you elaborate upon how you'd take advantage of DTMF A-D, how you'd produce the tones (are these standard now?), and what exactly you mean by muting from the far-end? DTMF A-D is not normally available by normal people. They're perfect for ADSI phones to use to initiate some kind of command since they do not get in the way of Joe's VoiceMail Service -- right now we seem to use * and # a lot, but so does everyone else. How do you escape these keys so that the far end can detect and use them? That's why I suggested using DTMF A-D to control asterisk with ADSI. I am fairly certain you can say Dial(Zap/1/D) and get the D tone I think. :-) It's be trivial to do if not, but I'm not so much looking at * to generate the tones as just detect them and have the ADSI phones generate them. Muting from the far end -- after reading it that way I think I see your confusion. :-) What I'd meant was that *, upon hearing one of these DTMF tones, mutes the channel so that the far end doesn't hear it, or rather hears a very (under 1/10s) short burst of it. It'd be both a security feature and a just plain nice feature, since when I'm transferring someone or calling up some feature on my ADSI phone while talking to someone, I'd prefer not to blast them with DTMF. :-) Hopefully that clears up what I'd been talking about. :-) Regards, Andrew ___ 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 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] MeetMe questions
Hi! - Can you limit the size of a conference room, ie max 8 people, etc. With MeetMeCount() and GotoIf() you are be able to limit the size of a conference room easily, it's a just a little bit of dialplan magic in extensions.conf. - Is there a list somewhere (besides the source ;) that has all the commands availible to people in the conferences? Specifically can you do a mute all new callers type action (when people are really just calling up to listen. http://www.voip-info.org/wiki-Asterisk+cmd+MeetMe - Passwords/Pins for the conference rooms? Use the dialplan. Put Authenticate() before MeetMe(). http://www.voip-info.org/wiki-Asterisk+cmd+Authenticate Cheers, Philipp ___ 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] FLASH TONE
Hi list. I'm having the next problem. I Bought a new analog phone, it have flash button, but it send a tone not a cut on the line. So the flash key is not working, a thing that was problem of the phone, but i connect another phone that have the same problem. I suppose that the flash key send a tone, becouse when i push it i lost the dial tone. Any idea how can i do, so * detect that tone as flash key ? Alvaro Parres ___ 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] Lucent and ISDN-PRI
Title: RE: [Asterisk-Users] Lucent and ISDN-PRI That document certainly is informative, thanks. I actually went with a tn464F that I happen to have and from the lucent side I have no problem setting it up as a signaling trunk group. Asterisk starts up, registers 1 D-Channel, and 23 B-Channels, but thats as far as I get. When I try to dial the asterisk via the Feature access code I defined on the definity I don't get any sign of a connection. The definity dials, and then waits until timeout at which point I get a busyback. Similarly, if I try to dial out from the Asterisk I get an all busy. I turned on pri intense debug span 1, to see if there were any obvious errors. When I do a dial I get the following traceback: start incredibly long debug message -- [ [02 [02 01 [02 01 01 [02 01 01 38 [02 01 01 38 ] [02 01 01 38 ] Supervisory frame: SAPI: 00 C/R: 1 EA: 0 TEI: 000 EA: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] N(R): 028 P/F: 0 0 bytes of data -- Restarting T203 counter -- Restarting T203 counter [ [02 [02 01 [02 01 38 [02 01 38 be [02 01 38 be 08 [02 01 38 be 08 02 [02 01 38 be 08 02 80 [02 01 38 be 08 02 80 f8 [02 01 38 be 08 02 80 f8 5a [02 01 38 be 08 02 80 f8 5a 08 [02 01 38 be 08 02 80 f8 5a 08 02 [02 01 38 be 08 02 80 f8 5a 08 02 81 [02 01 38 be 08 02 80 f8 5a 08 02 81 d1 [02 01 38 be 08 02 80 f8 5a 08 02 81 d1 ] [02 01 38 be 08 02 80 f8 5a 08 02 81 d1 ] Informational frame: SAPI: 00 C/R: 1 EA: 0 TEI: 000 EA: 1 N(S): 028 0: 0 N(R): 095 P: 0 9 bytes of data -- ACKing all packets from 94 to (but not including) 95 -- Since there was nothing left, stopping T200 counter -- Stopping T203 counter since we got an ACK -- Nothing left, starting T203 counter Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 33016/0x80F8) (Terminator) Message type: RELEASE COMPLETE (90) Cause (len= 2) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Invalid call reference value (81), class = Invalid message (5) ] Sending Receiver Ready (29) [ [02 [02 01 [02 01 01 [02 01 01 3a [02 01 01 3a ] [02 01 01 3a ] Supervisory frame: SAPI: 00 C/R: 1 EA: 0 TEI: 000 EA: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] N(R): 029 P/F: 0 0 bytes of data -- Restarting T203 counter -- Restarting T203 counter [ [02 [02 01 [02 01 3a [02 01 3a be [02 01 3a be 08 [02 01 3a be 08 02 [02 01 3a be 08 02 80 [02 01 3a be 08 02 80 f8 [02 01 3a be 08 02 80 f8 5a [02 01 3a be 08 02 80 f8 5a 08 [02 01 3a be 08 02 80 f8 5a 08 02 [02 01 3a be 08 02 80 f8 5a 08 02 81 [02 01 3a be 08 02 80 f8 5a 08 02 81 d1 [02 01 3a be 08 02 80 f8 5a 08 02 81 d1 ] [02 01 3a be 08 02 80 f8 5a 08 02 81 d1 ] Informational frame: SAPI: 00 C/R: 1 EA: 0 TEI: 000 EA: 1 N(S): 029 0: 0 N(R): 095 P: 0 9 bytes of data -- ACKing all packets from 94 to (but not including) 95 -- Since there was nothing left, stopping T200 counter -- Stopping T203 counter since we got an ACK -- Nothing left, starting T203 counter Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 33016/0x80F8) (Terminator) Message type: RELEASE COMPLETE (90) Cause (len= 2) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Invalid call reference value (81), class = Invalid message (5) ] Sending Receiver Ready (30) End incredibly long debug message -- Any suggestions? I feel like I am close.. but no cigar. :) Invalid message (5) anyone? I haven't looked at the libpri code but perhaps there is further explanation in there. I'm using pri_cpe channels 1-23, dchan=24, bchan=1-23. Any help is appreciated... thanks again, Matt -Original Message- From: Ken Godee [mailto:[EMAIL PROTECTED]] Sent: Monday, January 19, 2004 5:53 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Lucent and ISDN-PRI Matthew Branton wrote: Hi Everyone, So I have been further exploring the integration of our asterisk server and our lucent definity g3si system. I took the suggestion of setting up an isdn-pri line added the two way tie trunk and the signalling group, but can't seem to get the PRI signalling working on the asterisk correctly. I've set pri type to network on the lucent, and pri_cpe in zapata on the asterisk, but I am a bit confused as to the zaptel settings in this situation. It seems no matter what signaling mode I choose in zaptel.conf (with the exception of clear) I get an error on asterisk startup complaining about requested PRI vs unknown signalling. Any help would be appreciated in getting this working / ironing out some of my conceptual issues. :) I did get the lucent ot work under an em based tie group but that didn't seem to give me any more functionality than I had managed before. Thanks, Matt Matt, You know I'll be following this thread! Found a good reference for G3 isdn-pri you should
Re: [Asterisk-Users] MeetMe questions
On Tuesday 20 January 2004 12:28, Chris Robertson wrote: I'm looking into deploying * for an internal conference call server (using MeetMe) and had a couple of quick questions for those of you who have used it. I checked the Wiki but there weren't a lot of details for MeetMe. - Can you limit the size of a conference room, ie max 8 people, etc. Not directly, but you could run a MeetMeCount(confnum|varname), then check the results of varname and either allow/disallow that participant based upon the result. - Is there a list somewhere (besides the source ;) that has all the commands availible to people in the conferences? There really aren't any. Once you're in a conference, you can only exit the conference, if you entered with option p specified in the dialplan, by pressing a #. Otherwise, the only way to exit a conference is to hangup. Specifically can you do a mute all new callers type action (when people are really just calling up to listen. You want the monitor-only option, i.e. option m: MeetMe(1234|m) - Passwords/Pins for the conference rooms? For dynamic conferences, not yet, but you can with static-defined conferences: conf = 1234,4231 in meetme.conf. You can also pre-enter the PIN for a conference number in the dialplan: MeetMe(1234||4231). This might be useful if you wanted to protect the conference from people who could enter an arbitrary number but were using an alternate method of authentication. Sorry, there is currently no way to have multiple PINs per conference. Note that you do not currently need to restart or reload Asterisk for MeetMe to notice new entries in meetme.conf. -Tilghman ___ 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] Brandwidth for making internet calls
My ADSL connection speed is 512Kb up and 128Kb down. When making calls from Asterisk to IAX and back to the Asterisk, the sound is choppy and 20% of voice messages was lost. What is the production bandwidth requirement per internet call. I understand there is no guarantee of QoS but at least a benchmark to follow. -- David Kwok Iaxtel/FWD # 17001813482 smime.p7s Description: S/MIME Cryptographic Signature
Re: [Asterisk-Users] ADSI phone vs. IP phone
If I read above correctly, you imply that ADSI phones don't need wall-warts (A/C power transformers that plug into the wall). I'd assume that based on the sizable LCD screen, potential back-lighting, microcontroller(s), etc, that an ADSI phone would have to have a wall-wart, especially if you wanted to use any of its functionality while it is on-hook. I have designed a phone or two in my past (many years ago) and, as I recall, there is almost *no* current available from the telco while a phone is on-hook. You might be able to trickle-charge a very small battery, or run an RCA 1802 processor (microamps), but that's about it. Actually you are guaranteed a 20mA loop from the telco (and I would imagine from any channel bank as well) -- with CMOS technology you can do a _lot_ with 20mA... Unfortunately backlighting is not one of them, nor is driving a lot of LEDs. :-) And yes, I had not seen the forest for the trees and I ran smack into the middle of one of them. (I have done this in real life too once... the looks people give you...) I take back my statement that the wall-warts are an IP phone-only thing, this is simply not true. And IP phones give you the option of POE, something that you can't do with ADSI phones. Regards, Andrew ___ 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] MeetMe questions
I'm looking into deploying * for an internal conference call server (using MeetMe) and had a couple of quick questions for those of you who have used it. I checked the Wiki but there weren't a lot of details for MeetMe. - Can you limit the size of a conference room, ie max 8 people, etc. - Is there a list somewhere (besides the source ;) that has all the commands availible to people in the conferences? Specifically can you do a mute all new callers type action (when people are really just calling up to listen. - Passwords/Pins for the conference rooms? Thanks all, Chris Robertson Network Engineer Instill Corp. You could do all this through the dialplan fairly easily. I have already implemented everything you're talking about for several customers. - use MeetMeCount to deny additional users past N members - type show application MeetMe to get a list of commands - type show application Authenticate for password logic JT ___ 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
