Re: RE : Re: [Asterisk-Users] Multiple MWI on a single phone?
[EMAIL PROTECTED] wrote: We use the GXP-2000 and it works quite well. When you program the buttons to the individual lines, the message waiting light will light when any of the lines have a message waiting. When you press a line button, if there is a message waiting on that line, you will get stutter dial tone as well as a text message indicating the number of new messages waiting... I've been really pleased with these units. Ok, thats kinda how I thought it would work. So there is no way to visually indicate on the phone (ie turn on a LED) besides the dedicated MWI LED for each mailbox that has a message? Say I have a phone that I want to show 3 mailboxes. What I would like is to have three individual LEDS that show the status of each mailbox. Is there any phone out there that will do that? I've heard that the Cisco's will but those are expensive and you have to sell your soul to Cisco :-) Chris ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Multiple MWI on a single phone?
Hey all...I'm trying to find a phone that will support multiple MWI so that I can have a shared central phone with say 4 users who can see visually that hey have messages waiting. Is there any phone that will do this possibly by re-assigning a soft-button? Can the Polycoms do this since those seem to be the phone of choice these days? Thanks! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Multiple MWI on a single phone?
Bill Wesson wrote: Chris, I'm testing a Grandstream GXP-2000. It supports multiple MWI. Very nice! I didn't know about thatis there anything specific you have to do to associate a softbutton with a particular extension's voicemail so the MWI works? I didn't see anything about this in the wiki but I am looking over the user manual. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Answer confirmation on non-Zap channels?
I wholeheartedly endorse the idea of making this more generic and not channel specific...as to your ideas, I would be happy with having to press '#' to indicate acceptance of the call, even if there is only silence on the other end. On the other hand, I like your ideas of announcing the call, verbalizing the callerid digits and presenting a menu to answer or deflect, etc...that would be icing and very cool... So I see that this bug has been closed but it looks like it was due to no response from the patch owner. *IS* there a way to do call confirmation/acceptance with # on a non-zap (like IAX) channel or has this been dropped? -- Women like silent men, they think they're listening. http://ccicolorado.org Exceptional Dogs for Exceptional People - Help Out Today! ___ 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] Blank faxes with RxFAX
Patrick J. Conroy wrote: I wasn't able to get debugging information the first time around either. After pulling the latest asterisk from CVS, I was able to build and see debugging information when I started asterisk to test using asterisk -vvgc. But I noticed today that I do not get the same debugging information when I started asterisk using safe_asterisk. So, I don't know that rebuilding asterisk did any good. If you started asterik using safe_asterisk, I would shut it down, restart using asterisk -vvgc from a shell prompt and it may give you debugging information from the CLI. Thats right! I have seen that before...I'll see if I can start testing over the next few daysthank you! -- The Holocaust was an obscene period in our nation's history. I mean in this century's history. But we all lived in this century. I didn't live in this century.-- Vice President Dan Quayle, 9/15/88 http://ccicolorado.org Exceptional Dogs for Exceptional People - Help Out Today! ___ 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] Blank faxes with RxFAX
Patrick J. Conroy wrote: Hello All, I have finally pulled CVS HEAD and built it with app_rxfax and app_txfax to try to solve the problem that I was having with blank faxes. Fortunately, I am finally getting logs from rxfax. Unfortunately, I am still not receiving faxes correctly. Here is the log that was produced. If anyone has any thoughts on what might going, I would greatly appreciate it. I'm actually getting *some* blank faxes too...it seems that I can receive from an older crappy fax machine but not from a newer one like an HP all-in-one...how do you get debugging information so I can possibly help with this problem too? -- Happiness Is Seeing Your Mother-in-law on a Milk Carton. http://ccicolorado.org Exceptional Dogs for Exceptional People - Help Out Today! ___ 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] Cheap (US$120 or less) SIP Phones
Todd at Teledynamics (see wiki page mentioned above) has been very responsive to email, and we did not need to sign up as a reseller to purchase the Uniden phones. Great!! I'll give him a call today and see if I can order one...this looks like a really nice phone for the price and given the reviews from other people I'm actually kind of excitedhow do people get new firmware updates? Is there a website? -- Procrastination is the art of keeping up with yesterday. http://ccicolorado.org Exceptional Dogs for Exceptional People - Help Out Today! ___ 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] Cheap (US$120 or less) SIP Phones
James H. Thompson wrote: Are there any online retailers that carry the Uniden UIP series phones? I did a quick Froogle search to no avail. See: http://www.voip-info.org/wiki-Uniden So you *must* sign up as a reseller to purchase one? What are your opinions/problems on the UIP-200? It looks like a pretty good phone for a reasonable price. -- perl -e 'print $i=pack(c5,(41*2),sqrt(7056),(unpack(c,H)-2),oct(115),10);' http://ccicolorado.org Exceptional Dogs for Exceptional People - Help Out Today!
[Asterisk-Users] Problem with * not detecting hangup on FXO and VM going into an infinite loop
Hi everybody... I'm having an odd problem with voice mail on a recent CVS of * where it appears not to detect a hangup on FXO and * will keep treating the call as new and continue leaving voicemails until the max has been reached. It will then continue trying to leave voice mails and basically makes the system unavailble to any further incoming or outgoing calls on that FXO..has anybody seen this and if so how do I fix it? I've looked around on google and the list archives and it appears that there are others with similar problems with most people believing it to be a configuration problem. Since I don't see any bugs that have been formally posted with this description I think it most likely is...can anybody help me determine which option would be causing this behavior? I assume its in zapata.conf? Thanks! Chris -- It's easy to sit there and say you'd like to have more money. And I guess that's what I like about it. It's easy. Just sitting there, rocking back and forth, wanting that money. http://ccicolorado.org Exceptional Dogs for Exceptional People - Help Out Today! ___ 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] Problem with * not detecting hangup on FXO and VM going into an infinite loop
If you where on google and saw the same questions, you should have been able to follow the rabbit further down the hole by playing with the links at the bottom of the pages google provided. You would have seen use rant regularly that you need to work just a tad harder to find the answer. It is there. It is consistently the same problem. You lack disconnect supervision. The only way to know the line has been hungup is to use progress detection and possibly tweak for your location. Thank you humoring a newbie...I realize that we're a dime a dozen on this list and there a trillion emails that go through here a day basically asking the same thing...You have answered what I couldn't seem to find and that * can't know about remote hangups if its a not digital line. FXO doesn't cut it in this case...However there was an email Jorge J. Ramirez S. that stated that if I do a Hangup() after all Voicemail access that this should also solve my problem..so I thank you both!! I'll be posting this back to the wiki if it solves my problem and actually post back to jaiger on #asterisk who was also having the same problem. I'm not usually a leech..just people don't ask questions about stuff I know :-) Thanks again! Chris -- It's easy to sit there and say you'd like to have more money. And I guess that's what I like about it. It's easy. Just sitting there, rocking back and forth, wanting that money. http://ccicolorado.org Exceptional Dogs for Exceptional People - Help Out Today! ___ 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] smallest phone
Tim Sailer wrote: Folks, I'm looking for a SIP or IAX phone for field techs to take with them when out on service calls. The regular desktop phones are just way too big. Is there anything like the size of a full-sized cell phone? Or smaller, not I doubt that... If a softphone is acceptable what about something like http://www.kauss.org/Stephan/ziaxphone/ Can't get much smaller than that :-) -- The older you get, the better you realize you were. http://ccicolorado.org Exceptional Dogs for Exceptional People - Help Out Today! ___ 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] Cisco 7960 SIP image (off-topic)
Heison Chak wrote: $8/yr from Cisco. Seriously? Wow from Cisco thats actually reasonable! ehhe -- I can picture in my mind a world without war, a world without hate. And I can picture us attacking that world, because they'd never expect it. http://ccicolorado.org Exceptional Dogs for Exceptional People - Help Out Today! ___ 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] Migrating home POTS VM to Asterisk VM
I'm working on migrating my home POTS phone system into an Asterisk PBX. Currently I have my FXO and FXS setup and working great. Zapateller is nice! I'm working on my voicemail now and currently have my 3 mailbox answering machine plugged into my FXS along with all my other analog phones. I'd like to slowly migrate away from the FXS answering machine in favor of Asterisk Voice Mail. Currently a caller can press *3 to leave me a message in my own mailbox on the FXS machine. Is there any way that I can make Asterisk monitor the line after the machine has picked up and if it detects a *3 dump the user into my Asterisk voice mail box but for everything else just let the user leave a message on the normal machine? It was suggested on #asterisk that I could possibly accomplish this with a conference and AGI script but I'm not sure how I'd go about doing that. Thanks for any suggestions! Chris -- If all the world is a stage, where is the audience sitting? http://ccicolorado.org Exceptional Dogs for Exceptional People - Help Out Today! ___ 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] Problem with flashing FXO callwaiting from FXS
Hey all...I can't seem to figure out *exactly* what needs to be done when I can't flash over to an incoming callwaiting call on FXO from an FXS card. Right now if I get a callwaiting call from the FXO and hit flash nothing happens. I've been over the archives and google and it appears that this is possible and I guess it can be done with *0 but I haven't found the config on how to glue all this together. If anybody can help out I'd really appriecate it! Thanks, Chris -- Since light travels faster than sound, isn't that why some people appear bright until you hear them speak? -Steven Wright http://ccicolorado.org Exceptional Dogs for Exceptional People - Help Out Today! ___ 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] tunnel iax via gnophone with ssh?
Fearghas McKay wrote: Don't use CIPE, it has holes in it and is breakable. Use IPsec either FreeS/WAN or if you are running 2.6 kernel you can use its IPsec stack. Or an appliance to provide tunnels if you don't want to fiddle with kernels. I have FreeS/WAN setup for my wireless...I guess its time to set it up so its available from the outside...it would solve some other problems I've been having too... Whats the deal with teh IPsec stack in 2.6? Is it just FreeS/WAN and I don't have to patch any more? -- Software is like entropy. It is difficult to grasp, weighs nothing, and obeys the Second Law of Thermodynamics;i.e., it always increases. http://ccicolorado.org Exceptional Dogs for Exceptional People - Help Out Today! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] can't get caller id?
C M wrote: hi, i can get callerid in my phone directly connected to the pstn line. when i cannoect it o * it doen't give me callerid. i have set usecallerid=yes in zapata.conf file/ whaat could have happened? Has it ever worked? I had soemthing similar with a new FXO card...as it turns out it was bad... -- Only in America...do we leave cars worth thousands of dollars in the driveway and leave useless things and junk in boxes in the garage http://ccicolorado.org Exceptional Dogs for Exceptional People - Help Out Today! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] tunnel iax via gnophone with ssh?
Hey all...I'm trying to use gnophone to connect to my asterisk box behind my firewall..I thought I could just setup a tunnel with something like ssh host.com -L5036:asteriskserver:5036 and just change my gnophone to connect to localhost:5036 but I never see anything happen on the asterisk server. I'm even trying this on the same network just in case there is something funky with NAT. Anybody have any ideas? I did notice that when I start gnophone I see iax.c line 654 in iax_init: Started on port 5036 Listening on port 5036 and it doesn't seem to matter what I do inside the config. Are these ports in some way hardcoded? If if they are can't I do something like above? Thanks! Chris -- http://ccicolorado.org Exceptional Dogs for Exceptional People - Help Out Today! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: tunnel iax via gnophone with ssh?
Reinhard Max wrote: Asterisk uses UDP, but ssh can only forward TCP ports. Ahhh something I completly missed...that makes sense because I tunnel lots of other things...Are there other protocols that are TCP instead? -- To be intoxicated is to feel sophisticated but not be able to say it. http://ccicolorado.org Exceptional Dogs for Exceptional People - Help Out Today! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Anti-Ex Girl Friend logic (was Re: [Asterisk-Users] Setcontext based on CID...)
I hate doing metoos but I tried to get ahold of Michael Baird and never got a responsedoes anybody have the AGI code that Michael used for his Anti-Ex Girlfriend as described below? Thanks! Chris is the AGI available? Uriel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]]On Behalf Of Michael Baird Sent: Saturday, September 27, 2003 6:37 AM To: [EMAIL PROTECTED] Subject: Re: Anti-Ex Girl Friend logic (was Re: [Asterisk-Users] Setcontext based on CID...) I do it through AGI, I send the call to an external perl script, check the called-from-id against a mysql database, then send the call back to a context based on a ruleset I use, call-approved/call-not-approved/no digits received. Each context having a different voice message, so that the caller will know the problem, it works very well. Regards MIKE Blatantly stolen from Mark's presentation: exten = 600/2565551212,1,Congestion exten = 600,1,Dial(Zap/9,15) exten = 600,2,Voicemail(u600) exten = 600,102,Voicemail(b600) If the Caller*ID matches the ex-girlfriend (2565551212), provide immediate congestion tone. Otherwise try dialing on Zap/9 for up to 15 seconds. If there is no answer send them to voicemail, preceeded by unavailable message. If the interface is busy, send them to voicemail with a busy message. Jeremy McNamara Matt McIntyre wrote: I was wondering if someone might be able to offer a suggestion to me about how I might go about dropping a caller into a context specific to their CID. For example, I would like to be able to dial Asterisk from a specific number (a mobile phone) and have it drop me into a context other then the one that normal callers receive that has more options tailored to things I might want to do. I assume that answer can somehow be used to do this but I thought I might ask the experts and see what they might have to say. Thanks in advance, (You guys are great) Matt ^ ! Matt McIntyre (KF4FGZ) ! Certified Novell Administrator ! (336) 334-1134 (Campus telephone) ! (336) 215-7199 (Mobile telephone) - Please note the change ! (336) 334-1134 (Facsimile) ! E-MAIL: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ! AIM: MixMANJaVa ! ICQ: 11956085 ^ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Only in America are there handicap parking places in front of a skating rink http://ccicolorado.org Exceptional Dogs for Exceptional People - Help Out Today!
[Asterisk-Users] Making PrivacyManager smarter?
I'm having a problem with Asterisk picking up the Zap/1 and thinking its a new call when instead I've already been on the phone talking to a person. This is not my ideal setup and currently I have just an FXO card and Asterisk is in parallel with my phone system instead of being in the front. I'm not sure if the problem would be fixed by adding an FXS card and putting my phones on to that or not. This is what is happening while I'm on the phone. For some reason Asterisk all of a sudden things a new call has come in. -- Starting simple switch on 'Zap/1-1' WARNING[524305]: File chan_zap.c, Line 4360 (ss_thread): CallerID returned with error on channel 'Zap/1-1' -- Executing Zapateller(Zap/1-1, ) in new stack -- Executing PrivacyManager(Zap/1-1, ) in new stack == Parsing '/etc/asterisk/privacy.conf': Found -- Playing 'privacy-unident' -- Playing 'privacy-prompt' Is there a way to NOT have asterisk go through my s,2 if CID is an error? Or maybe change my incoming so nothing happens if CID returns with an error? extensions.conf [incoming] exten = s/,1,Zapateller exten = s,1,NoOp exten = s,2,PrivacyManager exten= s,3,Dial,IAX/192.168.2.69/|10 Thanks for any ideas! Chris -- Instead of a trap door, what about a trap window? The guy looks out it, and if he leans too far, he falls out. Wait. I guess that's like a regular window. http://ccicolorado.org Exceptional Dogs for Exceptional People - Help Out Today! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Stuttered Dialtone for multiple extensions
Hey all..I'm looking to start with a single FXS card but with 3 extensions for VM purposes only. I'd like to know if there is a way that you can have different stuttering dialtones depending on which extension has a VM. For example If x103 and x104 have VM can there be a distinctly different stutter for each mail box and have them play back at once or back to back so that when you picked up the phone you'd know who has voice mail? As I get the funds I'll be replacing my analog phones with IP but even cheap IP phones only support one mail box indicator right? Thanks, Chris -- Black holes are where God divided by zero. http://ccicolorado.org Exceptional Dogs for Exceptional People - Help Out Today! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Questions about Zapateller and Privacy Manager
Hey all...I'm just getting my * setup and right now all I have is an FXO but no FXS. I wan't to get rid of telemarketers by having * pick up the phone if there is no CID present, give the caller the Zapateller tones and then ask the user to input their phone number via Privacy Manager (yes I realize that this won't get us any where given that I can't re-ring the phones without FXS but that'll be my next purchase) This is what I have in my extensions.conf: [incoming] exten = s/,1,Zapateller exten = s,1,PrivacyManager exten= s,2,Dial,IAX/192.168.2.69/|10 and I *think* this works...is there a good way of testing this besides waiting for those lovely telemarketers to call? It was suggested last night on #asterisk that I put s,2,PrivacyManager instead of s,1,PrivacyManager but this seems to make PM think that there is no CID present and it always picks up. I guess it also gives the error of WARNING[327697]: File pbx.c, Line 1754 (ast_pbx_run): Channel 'Zap/1-1' sent into invalid extension 's' in context 'default', but no invalid handler -- Hungup 'Zap/1-1' Can somebody tell me that I'm doing this correctly AND explain why I'm seeing this behavior when I do s,2,PM? I'm sure it just has to do with not quite understanding how extensions works yet. Thanks! Chris -- I hope that after I die, people will say of me: That guy sure owed me a lot of money. http://ccicolorado.org Exceptional Dogs for Exceptional People - Help Out Today! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X100P Config
you can also do an insmod -N wcfxo (the -N checks only the numeric part the of the module and not the extra stuff) David J Carter wrote: Thanks Rich, I am re-installing the base SuSE Linux system again and will try to install everything without doing any updates. I can't remember any updates being done, but these automated installs for numpties like me could do anything and I wouldn't know. I will let you know how it goes. Cheers Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]]On Behalf Of Rich Adamson Sent: 13 October 2003 17:12 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] X100P Config When I run "modprobe zaptel" I get the message that the zaptel.o was compiled for kernel version 2.4.20-4GB while this kernel version is 2.4.20-4GB-athlon. And fails. When I run "modprobe wcfxo" I get the message that the zaptel.o was compiled for kernel version 2.4.20-4GB while this kernel version is 2.4.20-4GB-athlon. And fails. That's a real common problem discussed several times in the list. The issue is that somewhere along the line you've upgraded the kernel binaries (probably RedHat's up2date), and the source code that was installed in your base system (probably header files only) are from an earlier kernel. You'll need to install the kernel source for the actual version you are running. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Atheism is a nonprophet organization. http://ccicolorado.org Exceptional Dogs for Exceptional People - Help Out Today!
Re: [Asterisk-Users] FXO on ATT broadband POTS line?
I guess I should be more correct..its comcast not ATT. I get the two confsed :-) So Comcast is our POTS telephone provider. Chris Hirsch wrote: Does anybody out there run * on an ATT broadband phone line? I'm not seeing any callerid and I can't tell if its ATT doing something funky or if its my setup. I do see CID on my normal phones Thanks, Chris -- The are no stupid questions but there are a LOT of inquisitive idiots. http://ccicolorado.org Exceptional Dogs for Exceptional People - Help Out Today! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FXO on ATT broadband POTS line?
Does anybody out there run * on an ATT broadband phone line? I'm not seeing any callerid and I can't tell if its ATT doing something funky or if its my setup. I do see CID on my normal phones Thanks, Chris -- The face of a child can say it all, especially the mouth part of the face. http://ccicolorado.org Exceptional Dogs for Exceptional People - Help Out Today! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Is there always data at /dev/zap/1?
Hey all..in trying to futher troubleshoot my caller id problem I'm looking at some past troubleshooting tips and this struck me as strange: If I cat /dev/zap/1 I *always* see data...no matter if the line is in use or not...is that typical? Just curious... Thanks, Chris -- People are not homeless if they're sleeping in the streets of their own hometowns.- Dan Quayle http://ccicolorado.org Exceptional Dogs for Exceptional People - Help Out Today! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problems with Caller ID on FXO
And a followup for some debug messages for ManxPower: *CLI -- Starting simple switch on 'Zap/1-1' NOTICE[262161]: File callerid.c, Line 238 (callerid_feed): Caller*ID failed checksum ERROR[262161]: File callerid.c, Line 192 (callerid_feed): fsk_serie made mylen 0 (-30) WARNING[262161]: File chan_zap.c, Line 4340 (ss_thread): CallerID feed failed: Success WARNING[262161]: File chan_zap.c, Line 4357 (ss_thread): CallerID returned with error on channel 'Zap/1-1' NOTICE[262161]: File app_dial.c, Line 502 (dial_exec): Unable to create channel of type 'IAX' == Everyone is busy at this time WARNING[262161]: File pbx.c, Line 1810 (ast_pbx_run): Timeout, but no rule 't' in context 'inbound-analog' -- Hungup 'Zap/1-1' I understand the problem with the IAX stuff since I don't have that quite configured yet Chris Hirsch wrote: I'm still seeing this: *CLI -- Starting simple switch on 'Zap/1-1' NOTICE[262161]: File callerid.c, Line 238 (callerid_feed): Caller*ID failed checksum NOTICE[262161]: File chan_zap.c, Line 4314 (ss_thread): Got event 2 (Ring/Answered)... My zaptel.conf: fxsks=1 loadzone = us defaultzone=us My zapata.conf: [channels] language=en context=inbound-analog signalling=fxs_ks usecallerid=yes echocancel=yes echocancelwhenbridged=yes channel = 1 Now I just realized that I haven't set up my sound card on this computer and I'm seeing WARNING[98311]: File chan_oss.c, Line 232 (sound_thread): Read error on sound device: Resource temporarily unavailable Is that necessary to somehow read the caller id burst? Eric Wieling wrote: On Fri, 2003-10-03 at 13:28, Chris Hirsch wrote: failed checksum NOTICE[262161]: File chan_zap.c, Line 4314 (ss_thread): Got event 2 (Ring/Answered)... I've only seen this message when using callprogress=yes and/or busydetect=yes. Set them to no. -- Do pediatricians play miniature golf on Wednesdays? http://ccicolorado.org Exceptional Dogs for Exceptional People - Help Out Today! -- If at first you don't succeed,then skydiving isn't for you http://ccicolorado.org Exceptional Dogs for Exceptional People - Help Out Today!
Re: [Asterisk-Users] Problems with Caller ID on FXO
And a followup for some debug messages for ManxPower (2nd time since I can't seem to get all the details :-) ): FXO Card Installed: [EMAIL PROTECTED] asterisk]# cat /proc/zaptel/1 Span 1: WCFXO/0 "Wildcard X101P Board 1" 1 WCFXO/0/0 FXSKS (In use) Analog Line: ATT Broadband coming over cable and then into standard phone lines. Caller ID on other phones works fine. Asterisk Version is Latest from CVS as of around 09:30 MST today: This is the output when I receive a call: *CLI -- Starting simple switch on 'Zap/1-1' NOTICE[262161]: File callerid.c, Line 238 (callerid_feed): Caller*ID failed checksum ERROR[262161]: File callerid.c, Line 192 (callerid_feed): fsk_serie made mylen 0 (-30) WARNING[262161]: File chan_zap.c, Line 4340 (ss_thread): CallerID feed failed: Success WARNING[262161]: File chan_zap.c, Line 4357 (ss_thread): CallerID returned with error on channel 'Zap/1-1' NOTICE[262161]: File app_dial.c, Line 502 (dial_exec): Unable to create channel of type 'IAX' == Everyone is busy at this time WARNING[262161]: File pbx.c, Line 1810 (ast_pbx_run): Timeout, but no rule 't' in context 'inbound-analog' -- Hungup 'Zap/1-1' I understand the problem with the IAX stuff since I don't have that quite configured yet.... Chris Hirsch wrote: I'm still seeing this: *CLI -- Starting simple switch on 'Zap/1-1' NOTICE[262161]: File callerid.c, Line 238 (callerid_feed): Caller*ID failed checksum NOTICE[262161]: File chan_zap.c, Line 4314 (ss_thread): Got event 2 (Ring/Answered)... My zaptel.conf: fxsks=1 loadzone = us defaultzone=us My zapata.conf: [channels] language=en context=inbound-analog signalling=fxs_ks usecallerid=yes echocancel=yes echocancelwhenbridged=yes channel = 1 Now I just realized that I haven't set up my sound card on this computer and I'm seeing WARNING[98311]: File chan_oss.c, Line 232 (sound_thread): Read error on sound device: Resource temporarily unavailable Is that necessary to somehow read the caller id burst? Eric Wieling wrote: On Fri, 2003-10-03 at 13:28, Chris Hirsch wrote: failed checksum NOTICE[262161]: File chan_zap.c, Line 4314 (ss_thread): Got event 2 (Ring/Answered)... I've only seen this message when using callprogress=yes and/or busydetect=yes. Set them to no. -- Do pediatricians play miniature golf on Wednesdays? http://ccicolorado.org Exceptional Dogs for Exceptional People - Help Out Today! -- If at first you don't succeed,then skydiving isn't for you http://ccicolorado.org Exceptional Dogs for Exceptional People - Help Out Today!
Re: [Asterisk-Users] asterisk and 3com
We have the 3com NBX100 here at work..if anybody has ANY info on integrating * and the NBX I'd LOVE to hear about it. I'm not very happy with the NBX but loving what I'm learning about *. Hi! Anybody have experience using asterisk and 3com voip systems? Miklos ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Do pediatricians play miniature golf on Wednesdays? http://ccicolorado.org Exceptional Dogs for Exceptional People - Help Out Today! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problems with Caller ID on FXO
Hey all...for whatever reason my caller id doesn't appear to be working. My setup is simple (Wildcard FXO and thats it) and I'm just expecting the Caller ID to show up on the console. I'm seeing this: *CLI -- Starting simple switch on 'Zap/1-1' NOTICE[262161]: File callerid.c, Line 238 (callerid_feed): Caller*ID failed checksum NOTICE[262161]: File chan_zap.c, Line 4314 (ss_thread): Got event 2 (Ring/Answered)... and looking in the code I'm assuming this is bad :-) If it matters ATT broadband is my phone service and its coming through cable. Caller ID does show up fine on my standard phones. Can anybody give me a clue on this? Thanks! Chris -- Do pediatricians play miniature golf on Wednesdays? http://ccicolorado.org Exceptional Dogs for Exceptional People - Help Out Today! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problems with Caller ID on FXO
I'm still seeing this: *CLI -- Starting simple switch on 'Zap/1-1' NOTICE[262161]: File callerid.c, Line 238 (callerid_feed): Caller*ID failed checksum NOTICE[262161]: File chan_zap.c, Line 4314 (ss_thread): Got event 2 (Ring/Answered)... My zaptel.conf: fxsks=1 loadzone = us defaultzone=us My zapata.conf: [channels] language=en context=inbound-analog signalling=fxs_ks usecallerid=yes echocancel=yes echocancelwhenbridged=yes channel = 1 Now I just realized that I haven't set up my sound card on this computer and I'm seeing WARNING[98311]: File chan_oss.c, Line 232 (sound_thread): Read error on sound device: Resource temporarily unavailable Is that necessary to somehow read the caller id burst? Eric Wieling wrote: On Fri, 2003-10-03 at 13:28, Chris Hirsch wrote: failed checksum NOTICE[262161]: File chan_zap.c, Line 4314 (ss_thread): Got event 2 (Ring/Answered)... I've only seen this message when using callprogress=yes and/or busydetect=yes. Set them to no. -- Do pediatricians play miniature golf on Wednesdays? http://ccicolorado.org Exceptional Dogs for Exceptional People - Help Out Today!
Re: [Asterisk-Users] OT: Creating documentation using a web interface
You betcha!! We use it at work and we actually have non-html people contributing and fixing typos and errors..its easy to set up and easy to use. Leif Madsen wrote: Hm http://www.interactivetools.com/products/docbuilder/ This looks kind of what I want, but I am looking for a free version of something preferably. I would pay for something, but at this time I am unable to because of commitments to having to pay for tuition (still in school). Getting closer though. .. back to the google :) Jac Kersing mentioned to me to try twiki. Has anyone else used this with success to create documentation? Installing now as a test platform. Thanks, Leif Madsen. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Only in America do banks leave both doors open and then chain the pens to the counters http://ccicolorado.org Exceptional Dogs for Exceptional People - Help Out Today!
Re: [Asterisk-Users] OT: Creating documentation using a web interface
Heheh..lets try that again...we use TWiki http://twiki.org at work..and I'm in the process of setting it up at home..its a GREAT way to document anything. It starts as chaos and then eventually builds up a well defined structure as you go along. Chris Chris Hirsch wrote: You betcha!! We use it at work and we actually have non-html people contributing and fixing typos and errors..its easy to set up and easy to use. Leif Madsen wrote: Hm http://www.interactivetools.com/products/docbuilder/ This looks kind of what I want, but I am looking for a free version of something preferably. I would pay for something, but at this time I am unable to because of commitments to having to pay for tuition (still in school). Getting closer though. .. back to the google :) Jac Kersing mentioned to me to try twiki. Has anyone else used this with success to create documentation? Installing now as a test platform. Thanks, Leif Madsen. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Only in America do banks leave both doors open and then chain the pens to the counters http://ccicolorado.org Exceptional Dogs for Exceptional People - Help Out Today! -- "I promise you a police car on every sidewalk." - M. Barry, Mayor of Washington, DC http://ccicolorado.org Exceptional Dogs for Exceptional People - Help Out Today!
Re: [Asterisk-Users] Newbie just starting out with *
So its sounds like I do have a clue then...can analog devices have their own extension and do call parking, and paging and all that? I assume the caller id gets passed from the POTS CO to the internal phones? So from my understanding I can get a TDM400P with one port now and upgrade to additional ports? Thats what comes with the dev kit right? Tell me more about Zapateller..is that the script that I've seen a description of that gives the fake line-disconnected tone? McAughan, Matt wrote: RE: [Asterisk-Users] Newbie just starting out with * Chris: I started using Asterisk for very much the same reason. To blast those telemarketers and to improve my knowledge of PBX and telco. You have got a good start for a newbie. Yes the Wildcard X100P will terminate the POTS CO in to your Asterisk Linux Box. Then you have to figure out how to get everything "internal" connected up to it. I have a TDM400P. It can be purchased with up to 4 ports. I purchased three to save money. That gives me three "internal" extensions to plug analog phones in to. I just use three cordless phones with the base stations plugged in near the Asterisk computer. We leave one phone in the kitchen, one in the garage, and one in the bedroom. They can call each other when we are too lazy to go get one another or too far to scream at each other. They can all also share the one "external" line. Asterisk has been wonderful using Zapateller to blast those damn predictive dialers. The Asterisk voice mail has been wonderful too as it sends the recorded message to me and my wife at work as an attachment to an email. Best of luck, Matt -Original Message- From: Chris Hirsch [mailto:[EMAIL PROTECTED]] Sent: Tuesday, August 05, 2003 11:30 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Newbie just starting out with * Hey all...I'm brand new to * and I want to convert my home into a pbx type setup. I've figured out that I want a Wildcard X100P to bring my single POTS CO into my Linux box. My problem is that I'm sure sure what I need to do to get my analog phones connected up into a structured phone system. It *looks* like I can go the route of the Cisco Analog - VOIP for about $100 on ebay. That will get me two analog devices on the system. If I have four analog devices (2 normal phones, 1 fax and one 4 phone cordless system) is this the best setup? Do I need the TDM10B with the Asterisk TDM Dev Kit or does that just let me do one analog phone into the system? When converting from analog to VOIP do I get all the same features that I would if I got a TDM400P (4 ports of analog devices)? As I said I'm new and I would LOVE any pointers, HOWTOs or any good advice from people who have already done something similar. This project started out because I'm tired of the telemarketers calling and it looks like this will be the best and most flexable way to get my phone system wired up. I'm interested in any opinions on any real VOIP phones for a house (assuming VOIP) is the way to go. I envision that I could have a phone in every room, be able to do an intercom, MOH so I can hear music in each room etcideas? Thanks for the help and your patience, Chris -- The fact that no one understands you doesn't mean you're an artist. http://ccicolorado.org Exceptional Dogs for Exceptional People - Help Out Today! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- I hope some animal never bores a hole in my head and lays its eggs in my brain, because later you might think you're having a good idea but it's just eggs hatching. http://ccicolorado.org Exceptional Dogs for Exceptional People - Help Out Today!
Re: [Asterisk-Users] Newbie just starting out with *
Nice...so mixing and matching IP and POTS is ok and common then? I do care what the IP phones cost but then again I'm a gadget freak..if it has a big enough "cool" factor I may be able to justify price. What exactly does the Cisco phone get me over a Grandstream? I have an NBX100 (3com voip) at work and while it has its nify features 3com really dropped the ball on some of the features. Like 2K just to make it a true IP phone and then it only works with NT and not a H.323 gateway..but thats a whole other thing that irks me. Can any of the IP phones play a wav file as a ring tone? What about using an LDAP directory for dialing? I'd LOVE that and that may even increase the "wife factor" too hehehe. WipeOut . wrote: My .02c...(Purely my own opinions).. Get the X100P and if you need 4 analog ports get the TDM400P.. I use an X100P and a S100U(2 analog phone cordless system) and then 4 IP phones.. Your choice of IP phones are.. "I don't care what it costs" - Go for Cisco.. "I do care what it costs but still want something stylish and feature packed" - Go for Snom200.. "Cost is very important cos the budget is tight but I still want a good usable phone" - Go for Grandstream 101 or 102.. I have 2 Snom200's and 2 GS 102's.. Hope that helps.. shout if you have more questions.. Just a note.. IIRC fax machines sometimes have some issues.. Later.. Hey all...I'm brand new to * and I want to convert my home into a pbx type setup. I've figured out that I want a Wildcard X100P to bring my single POTS CO into my Linux box. My problem is that I'm sure sure what I need to do to get my analog phones connected up into a structured phone system. It *looks* like I can go the route of the Cisco Analog - VOIP for about $100 on ebay. That will get me two analog devices on the system. If I have four analog devices (2 normal phones, 1 fax and one 4 phone cordless system) is this the best setup? Do I need the TDM10B with the Asterisk TDM Dev Kit or does that just let me do one analog phone into the system? When converting from analog to VOIP do I get all the same features that I would if I got a TDM400P (4 ports of analog devices)? As I said I'm new and I would LOVE any pointers, HOWTOs or any good advice from people who have already done something similar. This project started out because I'm tired of the telemarketers calling and it looks like this will be the best and most flexable way to get my phone system wired up. I'm interested in any opinions on any real VOIP phones for a house (assuming VOIP) is the way to go. I envision that I could have a phone in every room, be able to do an intercom, MOH so I can hear music in each room etcideas? Thanks for the help and your patience, Chris -- The fact that no one understands you doesn't mean you're an artist. http://ccicolorado.org Exceptional Dogs for Exceptional People - Help Out Today! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- "The contagious people of Washington have stood firm against diversity during this long period of increment weather." - M. Barry, Mayor of Washington, DC http://ccicolorado.org Exceptional Dogs for Exceptional People - Help Out Today!
Re: [Asterisk-Users] How to Asterisk
Really? Thats awesome!! Thats why I wish there was a wiki available...is anybody opposed to one? What if I was to setup one at my site? Would anybody use it? Martin Pycko wrote: If you have a full duplex sound card then you should have such console commands available: dial answer hangup If you have internet try to do: dial 500 you should get connected with Digium PBX over IAX. You can create your own mailbox extensions, leave a voicemail, check voicemail and try to do your dialplan in extensions.conf using diffrent applications. regards Martin On Tue, 12 Aug 2003, prakashmodak_74 wrote: Hello, I'm new user of asterisk. Can anybody pls tell me how to use asterisk or any detail how to link i installed Asterisk-0.4.0 on i810 onboard sound card with Redhat 7.1. when i type "asterisk -vvvc" i get *CLI prompt Prakash Get Your Private, Free E-mail from Indiatimes at http://email.indiatimes.com Buy The Best In BOOKS at http://www.bestsellers.indiatimes.com Bid for for Air Tickets on Air Sahara Flights at Prices Lower Than Before. Just log on to http://airsahara.indiatimes.com and Bid Now! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Two wrongs do not make a right, but three lefts do. http://ccicolorado.org Exceptional Dogs for Exceptional People - Help Out Today!
Re: [Asterisk-Users] Newbie just starting out with *
Ok I'm convinced..one last question will a dual PII-266 500Meg RAM have enough horse power? McAughan, Matt wrote: Chris: Try not to be so worried about sound card, analog (FXO/FXS), digital (ISDN, BRI, PRI) and what is available by connecting device. The channel drivers take care of making the devicesavailable to Asterisk. In turn Asterisk makes all the features such as voice mail, call parking, and conference bridges available to the channels. Itis a beautifuland flexible design. many thanks toMark!With a few exceptionsmost features will be available to all connection methods. Yes you can upgrade the TDM400P. My thinking was to get at least two ports on it when I purchased it originally so I could call phone-to-phone internally without using our only external phone line. That way I could learn to configure and use asterisk with out annoying my friends in family trying to call in. Zapateller does not stop telemarketers it stops the predictive dialers they use. Ever received a call and answered hello two or three times before you get a person? That is a predictive dialer loaded with a list of numbers dialing all of their phone lines as quickly as possible. It will do it more efficiently than a group of agents witha phone number list in hand. When you answerthe dialertakes a moment to diagnose the fluctuations in the voice.The dialermakes a determination if someone even answered and if so if you are a person or an answering machine. If you turn out to be what it thinks is a realpersonit must find anavailable agent. That is what causes the pregnant pause. It has to find someone since you turned out to be a real person. Now what Zapateller can do is answer the phone and play the SIT (special information tone). When the dialer hears this it thinks your number is no longer in service and hopefully removes your number from that companies list. The other thing it can do is just play the SIT tones to any incoming call not providing caller id. Just take the plunge, buy the equipment, play around and come back here when you get stuck, Matt -Original Message- From: Chris Hirsch [mailto:[EMAIL PROTECTED]] Sent: Tuesday, August 05, 2003 12:13 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Newbie just starting out with * So its sounds like I do have a clue then...can analog devices have their own extension and do call parking, and paging and all that? I assume the caller id gets passed from the POTS CO to the internal phones? So from my understanding I can get a TDM400P with one port now and upgrade to additional ports? Thats what comes with the dev kit right? Tell me more about Zapateller..is that the script that I've seen a description of that gives the fake line-disconnected tone? McAughan, Matt wrote: Chris: I started using Asterisk for very much the same reason. To blast those telemarketers and to improve my knowledge of PBX and telco. You have got a good start for a newbie. Yes the Wildcard X100P will terminate the POTS CO in to your Asterisk Linux Box. Then you have to figure out how to get everything "internal" connected up to it. I have a TDM400P. It can be purchased with up to 4 ports. I purchased three to save money. That gives me three "internal" extensions to plug analog phones in to. I just use three cordless phones with the base stations plugged in near the Asterisk computer. We leave one phone in the kitchen, one in the garage, and one in the bedroom. They can call each other when we are too lazy to go get one another or too far to scream at each other. They can all also share the one "external" line. Asterisk has been wonderful using Zapateller to blast those damn predictive dialers. The Asterisk voice mail has been wonderful too as it sends the recorded message to me and my wife at work as an attachment to an email. Best of luck, Matt -Original Message- From: Chris Hirsch [mailto:[EMAIL PROTECTED]] Sent: Tuesday, August 05, 2003 11:30 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Newbie just starting out with * Hey all...I'm brand new to * and I want to convert my home into a pbx type setup. I've figured out that I want a Wildcard X100P to bring my single POTS CO into my Linux box. My problem is that I'm sure sure what I need to do to get my analog phones connected up into a structured phone system. It *looks* like I can go the route of the Cisco Analog - VOIP for about $100 on ebay. That will get me two analog devices on the system. If I have four analog devices (2 normal phones, 1 fax and one 4 phone cordless system) is this the best setup? Do I need the TDM10B with the Asterisk TDM Dev Kit or does that just let me do one analog phone into the system? When conve
Re: [Asterisk-Users] list proposal
What about a wiki? I personally don't like forums because I prefer the content pushed onto me and then let Mozilla do its magic with sorting and stuff. When I do have a problem however I first do a google unless there is a wiki present. Nine times out of ten, when users can post their own answers and help evolve documentation online it helps everybody, keeps everything organized and is usually much more up to date. BTW that also means that the main developers don't have to worry about keeping documentation up to date :-) That alone should be worth it hahaha. Just my .02 Chris Steve Meyers wrote: On Sun, 2003-08-10 at 21:31, Steven Critchfield wrote: On Sun, 2003-08-10 at 21:25, Andy Hester wrote: Perhaps there is another way to cut down on increased traffic... Specifically, I would go back to the suggestion of a collaborative website for documentation. Collecting info and organizing into Howto's would reduce the number of times people ask the same questions. Also, the documentation could grow as quickly as the project. Unfortunately, I don't have a place to host it currently. Ideally, the list would just be for issues that aren't already addressed. Any one else interested in this? While it still needs to be done, the majority of those type questions will still happen as the newest users still don't use google until told to do so. I don't buy that. I think that people are much more likely to check out documentation linked to directly on the site than they are to utilize Dr. Google's resources. Even if you google, the results can be confusing. Also, some people aren't quite sure what question they need to ask, and some entry-level documentation would help that. Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- "In Italy for thirty years under the Borgias they had warfare, terror, murder, bloodshed -- they produced Michelangelo, Leonardo da Vinci and the Renaissance. In Switzerland they had brotherly love, five hundred years of democracy and peace and what did that produce? The cuckoo clock." -Orson Welles, script of _The Third Man_ (1949). http://ccicolorado.org Exceptional Dogs for Exceptional People - Help Out Today!
[Asterisk-Users] Newbie just starting out with *
Hey all...I'm brand new to * and I want to convert my home into a pbx type setup. I've figured out that I want a Wildcard X100P to bring my single POTS CO into my Linux box. My problem is that I'm sure sure what I need to do to get my analog phones connected up into a structured phone system. It *looks* like I can go the route of the Cisco Analog - VOIP for about $100 on ebay. That will get me two analog devices on the system. If I have four analog devices (2 normal phones, 1 fax and one 4 phone cordless system) is this the best setup? Do I need the TDM10B with the Asterisk TDM Dev Kit or does that just let me do one analog phone into the system? When converting from analog to VOIP do I get all the same features that I would if I got a TDM400P (4 ports of analog devices)? As I said I'm new and I would LOVE any pointers, HOWTOs or any good advice from people who have already done something similar. This project started out because I'm tired of the telemarketers calling and it looks like this will be the best and most flexable way to get my phone system wired up. I'm interested in any opinions on any real VOIP phones for a house (assuming VOIP) is the way to go. I envision that I could have a phone in every room, be able to do an intercom, MOH so I can hear music in each room etcideas? Thanks for the help and your patience, Chris -- The fact that no one understands you doesn't mean you're an artist. http://ccicolorado.org Exceptional Dogs for Exceptional People - Help Out Today! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
