Re: [asterisk-users] Replacing mpg123 with madplay under Solaris?
I'm running Asterisk 1.2.12.1 on a Solaris 10 box. I've built mpg123 but it doesn't want to play well under Solaris so I want to replace it madplay. I've edited app/app_mp3.c and res/res_musiconhold.c to change the calls for mpg123 to madplay with the appropriate options. I'm using a little older version of asterisk with Madplay, but is it not still configured the same way, through the.. musiconhold.conf ; ; Music on hold class definitions ; [classes] default = custom:/var/lib/asterisk/mohmp3/,/usr/local/bin/madplay -Q --attenuate=-5 --mono -R 8000 --output=RAW:- iron2 = custom:/var/lib/asterisk/iron2/,/usr/local/bin/madplay -Q -z --attenuate=-25 --fade-in --mono -R 8000 --output=RAW:- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Shielding of T1/E1 cables WAS RE: Pinouts for T1/E1 crossover
Essentially true, but the impedance of a T1 cable is different from Cat5 cables, which is one of the primary factors in limiting distance. Has nothing to do with the twists. Shielded vs non-shielded has to do with the environment, and how much electrical noise there is near the T1 cable. Nothing more, nothing less. I always love these discussions on cat5 vs T1 cable. cat5 is NOT T1 cable and if any telco/vendor tried to install it in my location I'd have them pull it and put in the proper cabling. T1 cable is not just insulated cable, the cable pairs are separately insulated, not just for enviroment conditions but to prevent cross talk. The only safe way to try to use cat5 cable as a T1 cable would be two runs of cat5, one for Tx and one for Rx. It is necessary for the Tx and Rx signals to be in separate sheaths to prevent cross talk interferance. guess that's just me thou. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk service crashes
Here is a ps aux of the services while the server is crashed. Does anyone see any service that would have a conflict with the asterisk service? H, maybe just me, but I personally wouldn't run anything but asterisk on my server, yet alone adding... cups server font server http server ftp server database server email server Just my 2 cents. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: Where to buy a T1 crossover cable for * and channel bank
Does anyone know where I can buy a 50ft crossover cable to connect my digium card -- I believe it's a T100P -- to my Adit 600. The one I have now works fine but I need a longer one. You probally won't have a problem but Cat5 ethernet cable is really not T1 cable, it will work but I'll only use at the least 22awg, individually shielded pairs and shielded jacks. If going over the ceiling etc. I'll add a plenum rated jacket as well. These people make custom cables and have done a good job for me in the past. http://www.stonewallcable.com/dept.asp?dept%5Fid=71 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 NICs on Asterisk box
On reboot sometimes my onboard gigabit nic gets eth0 and sometimes the pci 3COM gets eth0 and this causes havoc with another piece of SW I run. Is it actually ethx getting flipped or the ip addresses? ___ 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] asterisk lock up
Any thoughts? I see others on the list have had similar problems but haven't seen any solutions :-\ Thoughts, h Not a lot to go on, what's top showing when this happens? I suspect a run away process consuming all resources. Is it really asterisk? I had this kind of problem at least once a week and it was mpg123 gone berserk. Once I replaced mpg123 with madplay no more problems. Just something to check. ___ 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] mpg123 zombie processes ...
I had noticed that MOH's mpg123 processes are not killed when asterisk is killed. Eventually after many restarts I see many of these zombie processes eating up CPU. Any Idea how could I make asterisk to clean up these properly. Do yourself a favor and switch to Madplay instead of mgp123. ___ 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] can't make my PRI dial out
I added the line exten = 3701,1,Dial(Zap/g1/19173657597) Unknown Number Plan (0) '19173657597' ] -- Called g1/19173657597 I know we are moving forward. I didn;t get this last time I tried to dial. Try striping the 1 off and dial Dial(Zap/g1/9173657597) ___ 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] Low cost box for hosting Asterisk and at leastone TDM400p
Chuck Bunn wrote: Hi, Yes cheaper than that - do not get me wrong I love Dell hardware but I do not need an installed OS, CDROM, Keyboard/mouse, and floppy. Minus all those I can get in down to $299 at the Dell site (using there 'N' series Optiplex - alternate OS box). I really need it to be a smaller box at a much lower price point... Go for a compaq deskpro sff 1ghz, with all the things you don't need, hd,cdrom,floppy,sound,nic with three available pci slots. These can be found on ebay $100-$125. There small, quite and linux friendly and make a very nice little asterisk box. It ain't gonna get cheaper than that. ___ 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] mpg123 home music from stream
Matt wrote: Where is my MoH class? I understand what's being done here... but I don't see where that is.. like for meetme conferences, and being placed on hold and such... which file? musiconhold.conf But once you get it going, it doesn't work anyway. Would love to have someone prove me wrong. Asterisk stops MOH (closes the stream) when channel hangs up. This is great for all other MOH uses, but drops the mp3 stream and doesn't reconnect to streaming sever. (as noted in original patch/bug #413) I was streaming XM radio thru MOH via shoutcast. Unless someones fix this problem. ___ 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] mpg123 home music from stream
Henry Devito wrote: Hi Ken, This has worked fine for me for about 6 months, maybe I just didn't notice a problem. As far as I know there has been music playing when people are being put on hold every time. Ah but you might want to take a closer look. If you can, watch the active connections on your streaming server. When you first start asterisk, you'll see connections formed from ast to your streaming server. Test music on hold, all is working, cool. Listen to stream as long as you want, works great. Now hang up, wait about 30 secs. and watch the connections drop off your streaming server. Test music on hold.When you test you will still hear music, but you won't see any new connections back to the streaming server, you'll just be listening to a buffered loop that was streamed in previously. Last * ver. I tried was 1.0.3 and I have not seen anything in the change logs thru 1.0.7 Did you take look at patch/bug #413? this describes the above problem. Same as I'm having. http://bugs.digium.com/bug_view_page.php?bug_id=413 ___ 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] mpg123 home music from stream
Matt wrote: Really? I just tried it and WHEN it's working.. it is streaming.. and even when I hang up it keeps mpg123 up and running in the background. Yes, doing a ps/top will show mpg123 processes but watch the precentage of cpu usage die to 0%. Asterisk may at any time have several mpg123 processes running, but they may all be idle. The way asterisk interacts with playing mp3's is very specific in how it plays and then saves cpu cycles by stopping the mp3 from playing (when ever there are no active channels) but leaving the mpg123/process in memory. This can be show even with normal mp3 moh, by having several mp3's and hanging up your test call to moh, then several minutes later dial back to your test moh and you'll still be on the same song. asterisk stops mp3 play to conserve cpu cycles and restarts when called on. Watch the connections on your streaming server 30-180 seconds after hanging up your test call to moh. Dial back in and watch no new streaming connections are made. Listen to your moh and observe your actually in a buffered mpg123 loop. ___ 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] Searching the list archives
Nick Stein wrote: This is probably a stupid newbie question. Is there a way to search the list archives? http://www.mail-archive.com http://www.mail-archive.com/asterisk-users%40lists.digium.com/ http://www.mail-archive.com/asterisk-dev%40lists.digium.com/ ___ 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] MP3 stream for MOH
CJ Toma wrote: Any suggestions how can I get asterisk to play MOH (music on hold) a MP3 radio stream from the internet (http:// location) instead of a MP3 file in the mphmp3 folder? I tried putting default = quietmp3:http://www.waixwave.com/pacnet.pls instead of default = quietmp3:/var/lib/asterisk/mohmp3 but did not work got message NOTICE[25564]: res_musiconhold.c:309 monmp3thread: Request to schedule in the past?!?! Any suggestions how to get the mp3 stream work? Thanks. CJ http://www.voip-info.org/wiki-Asterisk Has several examples. ie.. http://www.voip-info.org/tiki-index.php?page=Using%20Slimserver%20for%20Music%20on%20Hold slimp3 = custom:/var/lib/asterisk/mohmp3-dummy,/usr/bin/mpg123 -q -s --mono -r 8000 -f 8192 -b 0 http://localhost:9000/stream.mp3 But once you get it going, it doesn't work anyway. Asterisk stops MOH (closes the stream) when channel hangs up. This is great for all other MOH uses, but drops the mp3 stream and doesn't reconnect to streaming sever. (as noted in original patch/bug #413) Unless someones worked on fixing this. I was streaming XM radio thru MOH. But for now, your better off just moving along. ___ 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] Asterisk Management API
Umar Sear wrote: Hi all, I am trying to write an application to monitor queues using the Asterisk Management API. So far I have had some level of sucess, basically reverse engineering the protocol and the event messages using ethereal etc. I know there are a couple of pages on the Wiki that attempt (no dis-respect to who ever did it as it has been a great help) to document the API and was wondering if there is more information available. Any pointers will be greatly appreciated. I hope to document my findings on the Wiki once I have definative information. Thanks Umar Not sure what your looking for but you can just parse the output of the following commands show queues,show agents ie == Action: command Command: show queues == Response: Follows jrq has 0 calls (max 1) in 'ringall' strategy (0s holdtime), C:137, A:0, SL:50.4% within 0s Members: Agent/3041 has taken 137 calls (last was 10 secs ago) No Callers mwq has 0 calls (max 1) in 'ringall' strategy (0s holdtime), C:127, A:0, SL:44.9% within 0s Members: Agent/3042 has taken 127 calls (last was 68 secs ago) No Callers shq has 0 calls (max 1) in 'ringall' strategy (0s holdtime), C:0, A:0, SL:0.0% within 0s Members: Agent/3006 has taken no calls yet No Callers rgq has 0 calls (max 1) in 'ringall' strategy (0s holdtime), C:0, A:0, SL:0.0% within 0s Members: Agent/3009 has taken no calls yet No Callers bfq has 0 calls (max 1) in 'ringall' strategy (0s holdtime), C:0, A:0, SL:0.0% within 0s Members: Agent/1978 has taken no calls yet No Callers erq has 0 calls (max 1) in 'ringall' strategy (0s holdtime), C:0, A:0, SL:0.0% within 0s Members: Agent/3033 has taken no calls yet No Callers dwq has 0 calls (max 1) in 'ringall' strategy (0s holdtime), C:39, A:0, SL:51.3% within 0s Members: Agent/3007 has taken 39 calls (last was 4234 secs ago) No Callers dhq has 0 calls (max 1) in 'ringall' strategy (0s holdtime), C:87, A:0, SL:50.6% within 0s Members: Agent/3011 has taken 87 calls (last was 219 secs ago) No Callers mgq has 0 calls (max 1) in 'ringall' strategy (0s holdtime), C:0, A:0, SL:0.0% within 0s Members: Agent/3025 has taken no calls yet No Callers joq has 0 calls (max 1) in 'ringall' strategy (0s holdtime), C:0, A:0, SL:0.0% within 0s Members: Agent/3028 has taken no calls yet No Callers lsq has 0 calls (max 1) in 'ringall' strategy (0s holdtime), C:106, A:0, SL:41.5% within 0s Members: Agent/3017 has taken 106 calls (last was 12 secs ago) No Callers dmq has 0 calls (max 1) in 'ringall' strategy (0s holdtime), C:0, A:0, SL:0.0% within 0s Members: Agent/3010 has taken no calls yet No Callers sgq has 0 calls (max 1) in 'ringall' strategy (0s holdtime), C:57, A:0, SL:50.9% within 0s Members: Agent/3008 has taken 57 calls (last was 4797 secs ago) No Callers bcq has 0 calls (max 1) in 'ringall' strategy (0s holdtime), C:0, A:0, SL:0.0% within 0s Members: Agent/1674 has taken no calls yet No Callers thq has 0 calls (max 1) in 'ringall' strategy (0s holdtime), C:0, A:0, SL:0.0% within 0s Members: Agent/181 has taken no calls yet No Callers default has 0 calls (max unlimited) in 'ringall' strategy (0s holdtime), C:0, A:0, SL:0.0% within 0s No Members No Callers --END COMMAND-- == Action: command Command: show agents == Response: Follows 181 (Tom Hill) not logged in (musiconhold is 'none') 1674 (Bill Carron) not logged in (musiconhold is 'none') 3011 (Danny Harrington) logged in on Zap/4-1 is idle (musiconhold is 'none') 3028 (Justin Orstad) not logged in (musiconhold is 'none') 3025 (Mike Gaglio) not logged in (musiconhold is 'none') 3007 (Derrick Wilson) not logged in (musiconhold is 'none') 3008 (Steven Greenlaw) not logged in (musiconhold is 'none') 3033 (Eric Ryan) not logged in (musiconhold is 'none') 1978 (Bill Fornville) not logged in (musiconhold is 'none') 3006 (Saba Horton) not logged in (musiconhold is 'none') 3009 (Rob Giannina) not logged in (musiconhold is 'none') 3041 (John Rowley) logged in on Zap/16-1 talking to Zap/41-1 (musiconhold is 'none') 3042 (Michelle Wilson) logged in on Zap/15-1 is idle (musiconhold is 'none') 3017 (Laura Sood) logged in on Zap/2-1 is idle (musiconhold is 'rock1') 3010 (David McBrayer) not logged in (musiconhold is 'rock1') --END COMMAND-- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:
Re: [Asterisk-Users] Zaptel.conf and multiple T1 woes
Ben Ruset wrote: I have two Digium cards. One is a TE405P quad T1 card. The other is a TDM40B (I believe) quad analog POTS card. We have two T1's. Both of them are split in half (half voice, half data. - Don't ask me, that's how I inherited them.) Voice traffic flows on the back 12 channels of the T's. Our provider has been telling us that they are only seeing one D channel active. This would make sense if somehow only the first T1 in the 405P was activated. zaptel.conf: span=1,0,0,esf,b8zs span=2,0,0,esf,b8zs fxoks=1-24 bchan=12-23,36-47 dchan=24,48 loadzone = us fxsks=49-53 and zapata.conf: context=from-pstn signalling=pri_cpe switchtype=national faxdetect=incoming usecallerid=yes echocancel=yes echocancelwhenbridged=no echotraining=400 group=0 channel=12-23,36-47 context=from-pstn signalling=fxs_ks faxdetect=incoming usecallerid=yes echocancel=yes echocancelwhenbridged=no echotraining=800 group=1 channel=49-53 I could be wrong but. Wouldn't the channel numbering follow more along these lines? That's assuming you said that you've got the first span up which would mean the TE405P is card 1, otherwise it could be card 2. card 1 = TE405P === span 1 = channels 1-24 span 2 = channels 25-48 span 3 = channels 49-72 span 4 = channels 73-96 card 2 = TDM40B === 1st port = channel 97 2nd port = channel 98 3rd port = channel 99 4th port = channel 100 Also, what do you mean by I inherited them ? Where did they come from? Are you moving them from another piece of equipment? If so, are you sure the second span even has a D channel? Maybe it was part of an NFAS group? ___ 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] MusicOnHold
MF Hulber wrote: I'm looking for a simple way to disable MusicOnHold in my environment. I'm not really interested in having it and it causes too many problems with hanging mpg123 processes and memory management errors. The problem is, so many other modules seem to depend on it. I can't just cause a noload of MusicOnHold and be done. Does anyone have a simple solution? A solution that doesn't require a recompile is preferred but I'll appreciate and listen to any. After having the same issues you're having, we installed and now use Madplay. Been about 3 weeks and have not had a single issue with moh since. We where averaging several problems a week. http://www.underbit.com/products/mad/ musiconhold.conf [classes] default = custom:/var/lib/asterisk/mohmp3/,/usr/local/bin/madplay -Q --attenuate=-5 --mono -R 8000 --output=RAW:- rock1 = custom:/var/lib/asterisk/rock1/,/usr/local/bin/madplay -Q -z --attenuate=-5 --fade-in --mono -R 8000 --output=RAW:- ___ 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] manager api - Async:True?
Asterisk 1.0.3 / TE410 / ISDN/PRI Zap channels As I understand it using the Async: True in an originate action is supposed do a Fast Originate originate a call from a channel to an extension without waiting for call to complete. I'm finding no difference using Async or not, calls always wait for completion before connecting to extensions. Don't know if I'm missing something or if this just doesn't work when using ISDN channels and the ISDN signaling is overriding completion? I'm trying to do exactly what this feature is meant to do, connect channel-exten before completion. Any ideas? ___ 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] manager api events (pri vs pstn)
Asterisk 1.0.3 TDM400P/TE410P Using originate() call progress Events normal progression on completed call Event: Newstate State: Ringing Event: NewState State: up On pri Zap channels call progress events will wait @ State:Ringing until call FAILS via timeout if number dialed is disco'd, out of service, etc. and produce a progression of . Event: Newstate State: Ringing (long boring wait) Event: Hangup Cause: 0 The only exception is if dialed number is busy, then will instantly go from ringing to... Event: Newstate State: Ringing Event: Hangup Cause: 17 So on pri Zap channels it seems there are only three causes that get issued on hangups 0 (not defined) 16 (normal clearing) 17 (user busy) On analog PSTN Zap channels every call goes directly from State: ringing to State: up regardless of call completion. Which allows calls to be transfered instantly and user can then disposition call accordingly. Our development system is using TDM400P and production system using a TE410P Am I missing something? or is Asterisk not reconizing the status on the pri Zap channel, or is it, and just not issuing event causes for them? Does anyone know if work has been continued on this, to pass proper cause codes and not wait for call FAIL in 1.0.x or cvs? Or is there anyway to get around this so calls procceed without waiting for a FAIL/timeout, much like PSTN Zap channels do? We need this, to the extent we may have to install multiple analog lines and shed our smarter pri line. Suggestions? ___ 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] T1 Timing Slips
Ken Godee wrote: Does anyone know how to monitor * to see if they are receiving timing slips on a span connected to a T100P card? I am seeing b-channel restarts quite often and also getting No D-channels available warnings from time to time. Yesterday I had all the b-channels crash during a MeetMe Conference. Not good! This PRI is connected to an Avaya Definity PBX that is onsite and located in the same room as *. * is set to clock off the Definity. I am seeing no problems on the PRI from the Definity side. Just thought I'd run this by you. We've been running connected to our Definity G3si R6 via TN767 -- TE410P and have had no problems. I guess I'll eat crow alittle bit I guess your email made me focus a little more into it. I'm also having the same problems as you are, D-Channel bouncing. D-channel down and right back up and then b-channels restart, while restarting they DO drop any active channels. :( Experenced first hand on friday while remote monitoring and on a call. D-channel down No D-channel found, using channel 48 anyway. D-channel up restarting channel etc. As another poster suggested, I tried changing timing to internal clocking, vs. Definty, no help thou. I've done a ton of searching and have not found much more I can try. What protocol are you using on the Definity side? As I understand it a = ni1 / b = national If you come across anything that helps, please let me know. I'll also let you know if I find anything. I also see no problems on the Definity side. No errors when loop up circuit either. ztmonitor runs 100%-99%. No missing interrupts, etc. Load/no load doesn't seem to make a difference. Running astersk v1.0.3 ken ___ 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] Streaming Audio - Music On Hold Feature
Justin Richards wrote: I have not used any M$ products, but it works with shoutcast like this: default = quietmp3:/var/lib/asterisk/mohmp3-empty,http://host.com:8000/ basically, create an empty directory to point it to first, then the url to the stream. If the microsoft stream can be played via url in winamp in MP3 format, then it should work about the same. Justin, How are you keeping the mp3 stream open? My mpg client connections are closing after about 30-105 secs. The moh/mpg processes remain running and moh works fine but they're just looping whatever has been previously streamed before connections dropped. Is this not happening on your system? It is doing this on v1.0.3 ___ 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] T1 Timing Slips
Does anyone know how to monitor * to see if they are receiving timing slips on a span connected to a T100P card? I am seeing b-channel restarts quite often and also getting No D-channels available warnings from time to time. Yesterday I had all the b-channels crash during a MeetMe Conference. Not good! This PRI is connected to an Avaya Definity PBX that is onsite and located in the same room as *. * is set to clock off the Definity. I am seeing no problems on the PRI from the Definity side. zaptel.conf is as follows: span=1,1,0,esf,b8zs bchan=1-23 dchan=24 defaultzone=us loadzone=us zapata.conf is as follows: switchtype=national overlapdial=yes signaling=pri_cpe group=15 channel = 1-23 usecallerid=yes callerid=asreceived musiconhold=default echocancel=yes echocancelwhenbridged=yes echotraining=yes Just thought I'd run this by you. We've been running connected to our Definity G3si R6 via TN767 -- TE410P and have had no problems. Our zapata.conf settings our just a little different, might give you something to try. We had some strange problems when set as you have above. switchtype = 5ess overlapdial = no signaling = pri_net ___ 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] moh mp3 streaming problem
asterisk v1.0.3, mpg123 v59r, shoutcast server. When first starting asterisk all is fine, moh/mpg processes start, can see asterisk client connections on shoutcast monitor as well and I've got mp3 streamed music on hold, cool! After aprx. 32-105 seconds the asterisk client connections close on the shoutcast server. The moh/mpg processes are still running, but are now just looping a buffer full? of previous mp3 streamed music. asterisk MP3Player works as expected. mpg123 works fine from console, xmms too, etc. Moh seems to have some type of time out. Nothing in logs. I know there's other people streaming MP3's to moh, is this happening to you? I've tried to peek thru the res_musiconhold.c file but just can't figure it out. Class doesn't seem to matter, mp3,custom, even httpmp3. Any ideas how I can keep the MP3 stream open? (hope I made sense) ___ 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] Definity PBX with a T100P TN767E
I'm currently playing with a Digium T100P card and 2 Grandstream phones, things are working well. I wanted to move on to linking our Definity G3R Rev 8.2 to the T100P. Everything that I've read so far shows that you need a TN464 to accomplish this. We have a TN767E available. Yes, a TN767E will work and actually a TN464 may not, depending on how the G3 is setup. If I remember right the TN464 needed a different clock set up then we had on our system. I've got my G3 working with asterisk using a TN767E (v18 R11 - Ebay $100, gotta love Ebay). Inbound/outbound, DID from G3 inbound, ext./ext., etc. You just have to make sure your G3 has a spare proc. interface and of coarse you already have PRI ($ feature enabled) on the G3, right? Here's some notes from when I did mine, hope they help you. Hotplug cp (purple slot) in spare slot, ie. 01A06 add DS1 01A06 display DS1 01A06 add data module with type of 'procr-intf' and a non-DID extension number Assign the data module to a physical channel (01 to 04) Do a 'change communications-interface links' to add the information for the ISDN board. Use the same physical channel as assigned to the data module. Enable = n Est Conn = y PI Ext = Data mod created above PROT = ISDN Brd = TN767 slot Identification = whatever Do a 'change communications-interface processor-channels' and add an entry: Appl = ISDN Link = same as assigned to the data module Channel = blank Priority = h Do a 'add or change signaling-group x' Associated Signaling = y for facility associated sognaling n for non-facility associated signaling Primary D cahnnel - 767 slot, port 24 Trunk Group = ? Go back to the 'change communications-interface links' form and enable the link that you are using. Give it a few minutes to sync up and then do a 'status signaling-group x' You should see the primary as 'in-service' ___ 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] handset to sound card
Norberto Harmath wrote: Does anybody know how to build a handset to sound card adapter ? Might try looking here. http://www.sandman.com handset for soundcard http://www.sandman.com/serial.html I think there's more, just couldn't find. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] (Analog Intercom) PagePal by ATT -- was hooked to a Merlin
Jeb Campbell wrote: I'm replacing a Merlin for a client and they have a PagePal Intercom that I would like to reuse. Here is what I know about it: It has a screw-down wires that goto rj-11 (This was told to me over the phone) that went into one of the Merlin ports. I tried bring it up with fxo_ks and fxo_ls (assuming it was analog and autoanswered) but no luck. I would be happy to replace if anyone knows of an analog phone to page system, but of course I would like to reuse what is there. Thanks for any advice or pointers, Don't know about your PagePal unit, but we've been please with the Valcom units. They also have some interesting Voip page units available that are pretty cool. http://www.valcom.com/oneway.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] Avaya/Lucent Definity - Asterisk interop question
The box has a T100P card hooked up to a csu on the Definity with a patch cable. A. You don't need a CSU if located close to each other. B. Patch Cable? If using CSU, straight thru cable, if no CSU cross-over cable. C. Make sure to know your pin outs on the CSU vs pin outs T100P Lucent/Avaya/ATT or whatever, sometimes like to screw with us and use non standard configurations for pin outs. Where as in a normal situation a straight thru would work, might not if the pin outs are different. Just a couple of thoughts. Steve Kann wrote: I'd really like to figure out a way to map a set of extensions on the definity to automatically be handed off to asterisk. For example, have all extensions in the range 4900-4999 end up being calls to asterisk with the extension number. Create route pattern ie. route 14 = asterisk trunk group dialplan eta routing pattern = 14 digit 4, length 4, extension When someone dials ext 4900-4999, and it's not defined locally, it will follow the eta to asterisk. This will send all numbers dialed that are not defined locally thru the eta, but that's no big thing. There is a lot of ways to do some of this stuff, some are also dependent on what Definity options are enabled. Also, certain security settings. I've got the following working (by hook or crook) Asterisk ext. - Definity ext. Asterisk outbound via Definity/ars Inbound DNIS Definity - Asterisk ext. Here's a tricky one, seems sloppy but works... Inbound Definity/auto attendant - Asterisk ext. Transfering inbound call out of Audix and then off switch, security guys don't like that to happen. ___ 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] Connecting Asterisk and Avaya Definity By E1 . Incoming work, but not outgoing
Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Incompatible destination (88), class = Invalid message (5) ] Here's how I've got mine set up, maybe it will help, it's a little different then how the wiki has it. I'm running Definity G3si v6 (ISDN PRI) TN767E v18 -- TE410P -- zaptel.conf -- span=2,1,0,esf,b8zs # span 2 bchan=25-47 dchan=48 loadzone=us -- zapata.conf -- ; isdn-pri - att pbx group = 3 immediate = no switchtype = 5ess overlapdial = no signalling = pri_net channel = 25-47 ___ 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] Connecting Asterisk and Avaya Definity By E1. Incoming work, but not outgoing
Roman Bessyadovskii wrote: Hi All. I connect asterisk and definity by manual at www.voip-info.org/tiki-index.php?page=Asterisk%20Avaya. (I just only have E1, not T1 card). I see, that card work (in definity trunk status, and at asterisk Incoming call, from definity is work ok, but when I try outgoing call, I recive -- Executing Dial(SIP/1015-870f, Zap/g1/2073) in new stack -- Called g1/2073 -- Channel 1, span 1 got hangup -- Hungup 'Zap/1-1' == No one is available to answer at this time How fix it? Do you have the Dial Plan set up properly on the Definity side? ___ 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] * as pri_net?
Bruce Komito wrote: If you have used * to support a pri as pri_net (as opposed to pri_cpe), either to talk to another * system or a PBX of some sort, I would be very interested in hearing about your experiences. Imparticular, I would like to know that it works before I invest in the extra hardware. TIA Bruce Komito Here too asterisk/TE410P ISDN-PRI TN767E/Definity G3si v6 switchtype = 5ess signalling = pri_net inbound/outbound, ext/ext, DNIS/ANI all working well. Very cool! ___ 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] T100P T1 problem (Avaya - asterisk IVR)
Michael Welter wrote: Actually, please leave this thread on the list. Question: since this is a local connection between the Definity and Asterisk on a crossover cable, could E1 PRI be used, even though we're in the US, to realize another 8 channels? I have TN464F cards that I will be using to connect with Asterisk. Thanks, Mike James Coberly wrote: Jeb, Do you know what slot it is in? Carrier A (top) or B (bottom)? We should take this off list though and reply to me directly from this point, since this is not really * related now. There are 2 ways to do this: At the system propmt type: list configuration ds1 (will list all DS boards in the system) list configuration all will give you all boards in the system. FInd the one related to the slot you are connected to. Or if you have a restricted shell: You can look at the back of the unit, locate the amphenol you connected, there is a no. (slot #) Locate the card on the front of the unit in that slot. Should be marked TNXXX James- Yes, leave on list, or someone cc me, have exact same project definity TN767E - * coming up very soon so like to follow progress. ___ 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] small correction
kemal asad wrote: as i am trying to use asterisk and install my newly purchased ( got it yesterday) digium cards. i am following the very detail steps of http://www.automated.it/guidetoasterisk.htm. but one thing did not seems right so i wanted to let enveyone know the page says: Once compiled make sure there is a copy in /usr/bin/mpg123 i think the location is /usr/local/bin/mpg123 Redhat has the mpg123(mpg321) binary in the /usr/bin/ other distro's may have it some where else. I believe the point, is to make sure you have a copy in your PATH statement. Also as side note(common problem), make sure you really have mpg123 and not a symbolic link to mpg321 . I think it's a Redhat thing and sounds like you're on a different distro anyway. ___ 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] Asterisk 0.7.1 RH 7.3 RPMS Released
This is great to see.. but why RH7.3 (or RH8 for that matter) since it has already been EOL'ed by RH?? Couple of reasons.. 1. It is a stable, known quantity that uses solid components and closely mirrors the environment that a lot of people develop Asterisk on. It isn't going to drastically change, so those wishing to deploy it in production may look to RedHat 7.3 as a stable platform for that purpose. I agree, keep up the good work. I personally don't see any reason to upgrade atleast until the 2.6.x kernel is well underway. Maybe that's just me, hell I'm still running a 4.11 Novell server and a SCO Open server that hasn't been touched since y2k upgrades. Also if you look around for stable/available drivers from manufactures you'll find mostly 7.3 and some 8.0 supported drivers. Just try to call a manufacture and tell'em your having problems running their hardware with the newest greatest version of x.x.x, but if you're using one of their supported drivers you'll get the support you need. So moral of the story, always check with the hardware manufacture and stay with supported distributions. Just my .02 ___ 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
Matthew Branton wrote: 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 -- Here's some more reading and also a great refernece to have (Just as soon as I can figure it out, that is) You can go to the ITU electronic book store and download up to 3 recommendations free of charge, I'd grab Q.850 , Q.921, Q.931 You have to register first at... http://ecs.itu.ch/cgi-bin/dms-ebookshop then download from here http://www.itu.int/rec/recommendation.asp?type=productslang=eparent=T-REC-Q From the little I understand ISDN Malfunction (81)Invalid Call reference (out of parameters) (5)Miss Dialed trunk prefix Someone should know what all this means, I really don't have any idea, Still digging myself, through tons of info/implemention manuals/etc. Just getting ready to purchase my TN767 or TN464. ___ 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
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 have a look. Go's into good detail, more then the stock implementation manuals do. http://support.avaya.com/elmodocs2/multivantage/025107_1/025107_1.pdf I noticed you said you where going to use a TN767(E) circuit pack, did you? Depending on your system(version) you might have to use TN464 for ISDN-PRI It's questionable as to weither you can use a TN767 for ISDN-PRI w/FAS Some systems you can, others you can not. You might need to install a packet adjunct TN555 Are you sure your D channel is up? It's seems the TN767 is mostly used as ISDN-PRI w/NFAS. Anyhow just a thought, cause that's where I'm at right now is trying to figure out which circuit pack I need to use. ___ 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] Zone Paging
There are a number of paging interfaces available which connect to a regular phone line on one side and to a paging amplifier on the other side. Could you provide a pointer? The search terms pager and telephone together are giving me a heck of a lot of noise. . . Thx. B. http://www.valcom.com Easy to use, widely available and fairly inexpensive. ___ 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] Asterisk Integration with Lucent Definity g3si
Matthew Branton wrote: Come monday I will see if I can get the PRI line working if we have an extra 767 circuit pack. I promise that if/when we get this working I will definitely write up a detailed explanation of the steps involved. Right now we have a partial setup but a fully integrated box seems within reach... any more specifics would be great. Matt Matt, Been doing a little digging, so far found some good info on Cisco's web site, here's an example.. http://www.cisco.com/application/pdf/en/us/guest/products/ps259/c1237/ccmigration_09186a00801475be.pdf (The above url probally got split) Also go to Cisco's web site and do a search on G3 migration Lots of info on migrationg legacy G3 to voip and also some good war storys. There's also a very good definity forum that's very active with many Definity consultants willing to give a hand on all aspects of G3 system admin. http://www.tek-tips.com/gthreadminder.cfm/lev2/9/lev3/89/pid/690 Sounds like you're a step ahead of me on this, but the above forum is a good stop as soon as I dig through the manuals a little and brush up on my G3 talk. Let us all know how it goes, I'm sure it will work fine and just as soon as I can get to it, I will get it going. 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] Asterisk Integration with Lucent Definity g3si
PBXtech wrote: We have our G3R setup on a PRI connection. Your trunk group should be set to tie. If anybody would be willing to share just alittle more info on how to set this up it would be great. I've just started thinking about this also. A brief outline would be great, circuit packs used, ds1 settings, trunk group settings and how are you guys setting up the private network to route calls through to/from * ? There's just not a whole lot of info out there on this and I'd rather cut my left nut off, rather than try to talk to Avaya about this. I noticed a spot on the wiki for just this thing, but no ones contributed. http://www.voip-info.org/wiki-Asterisk+legacy+integration Either it's simpler or more complex than I'm making it * via TE410P ISDN-PRI - G3 ISDN-PRI DS1 TN767E (dependancy circuit packs in place, via existing ISDN-PRI) define G3 DS1 assign (tie) trunk group to the G3 DS1 A.)How should one route calls through the G3 out to extentions defined in *? B.) How should one route calls from the * server to/through the G3's ext.s and outbound lines? Any info would be great, ___ 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] 3.3v PCI board - TE410P photo
mattf wrote: No need to go Xeon, I have one of these: http://www.tyan.com/products/html/thunderk7x.html Dual AMD Athlon MP with one 3.3v 64bit PCI slot MATT--- Athon MP or Xeon IMO are the same thing.. They are just the high end version of either the AMD or Intel proc respectively.. Later.. Ok, so I'm a compaq kind of guy but I can't even remember off the top of my head any of their servers that don't include 64-bit 3.3v slots, even the lower end, older G2, Pentium III based servers. ie. ML350/G2 PIII 1.26ghz includes... 64-bit/33MHz,PCI(5 available) 3.3 Volt ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] analog or sip ? was far end disconnect supervision
On Sun, Jan 11, 2004 at 05:37:55PM +, Miguel Cavazos said: Sip phones get old and look ugly, analog can be replace at any moment. Frankly, *good* analog phones cost almost $200. If you want anything with features (such as ADSI) it's gonna cost as much as a good SIP phone. If someone Does know of a good analog phone that has good speed dialing, good headset support, a decent display, good sound quality, and is reasonably priced I'd be very interested (no ebay - I want new.) Been pretty happy with the Aastra 480 w/ADSI @ $124.00 http://www.twacomm.com/Catalog/Model_PT480.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] Programming an unlocked ADSI phone?
Darren Nickerson wrote: That worked a treat - thanks! Comedian Mail is now able to download to the handset and there's a lot more functionality now. -d I'd be interested in knowing if once you try to use Comedian mail softkeys if the 480 keypad goes dead? Mine and several others reported same, which makes it useless, a shame to, I like the 480's ADSI function and haven't had a whole lot of time to look into it. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] is there any way to search the mailing list archive and order results by date?
http://www.mail-archive.com/asterisk-users%40lists.digium.com/index.html Returns searches in chronological order. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Uptime counters (was RE: [Asterisk-Users] Bayonne and Asterisk)
James Sizemore wrote: I did not even know about it! But seeing as it is not in the change log no wonder? You have the bug number the notes are under for usage? ID # 345 10/02/03 - logger_reload.diff Summary - 'logger reload' CLI command Description - Closes and reopens the log files. Good for those wanting to rotate log files. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Uptime counters (was RE: [Asterisk-Users] Bayonne and Asterisk)
Andrew Kohlsmith wrote: You can not rotate logs with out dropping calls, and if logs get a little over 2Gbs Asterisk will crashes... Why not? Why are the logfiles kept open for the entire life of Asterisk? Hell even my heavily loaded qmail server isn't this braindead in that regard. Maybe I missed part of this thread, but as of like 10/05/03 cvs there was a new app added for this called (I think) logrotate. It's supposed to allow you to send * a remote command and rotate your logs. I upgraded for this feature but have not had time to test it yet, it's on my look at list. Like I said maybe I missed part of thread but you should be able to setup a cron job and forget about it. Anybody using the logrotate app? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Asterisk Lists (was Re: [Asterisk-Users] Asterisk Business discussion again)
Mark Spencer wrote: Amen! While -dev and -users may be a little too sparse, perhaps adding a -business list would be beneficial for discussing those types of issues. However business-related issues are not so common at this point, so perhaps a list devoted to NONTECHNICAL discussion (-nontech?) would be relevant? I agree that list fragmentation is a royal pain in the ass, but perhaps it is time to figure out just one more list to try and whittle down the traffic on -users. So far it seems like the proposed candidates for new lists are: asterisk-newbies (perhaps a better word?) asterisk-nontech asterisk-biz Any others as well? If we were to add another list, I *believe* we could automatically subscribe everyone in -users to -whatever to help seed it a bit. asterisk-newbies bad idea, been tried many times, who's going to subscribe to that to try to get answers. It's important the newbies get help from people with the knowhow (if they want to help them). Not just avoided, besides that they'll just join the users list anyway and ask the question again. I'm on a couple high volume list (python/qmail) and I hate to say it but, the best ways I've seen to keep posts down are. 1. A link to guidelines for posting to the list ie. http://www.qcc.ca/~charlesc/writings/12-steps-to-qmail-list-bliss.html Instead of someone coming accross wrong, you send them to a link like the above. 2. Having a couple of guys around that don't mind coming accross a little brash. It's sets the feel for the list and people WILL spend more time researhing it before writing the list. Hell, I've been told many times to RTFM, google it, etc. I guess I'm just not that thin skinned, and because of it, that's what I've learned to try to do first. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] manager.conf
Steven Critchfield wrote: On Tue, 2003-11-18 at 12:37, George Lin wrote: Hi, Do you know if we can use AGI or other script to handle the asterisk events by using the existing asterisk manager process ? AGI is for handling calls. AGI is to phone calls like CGI is to web page requests. There is a perl module to use in accessing manager events though. search the archive for links to it. And also a very good python module available .. http://sourceforge.net/projects/pyst/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] mpg123 causing Asterisk Freeze?
mattf wrote: Hello, I am currently using MusicOnHold(mpg123), and it works just fine, but every once in a while I will get a flurry of warnings in the CLI like those below and Asterisk will freeze completely, and the only way to come out of it is with a kill -9 . Is mpg123 causing my problem? Is there a specific format of MP3 that should be used/avoided to not have errors like these? Any help would be greatly appreciated. Thanks, MATT--- ERRORS: Skipped RIFF header! Warning, flexibel rate not heavily tested! Warning, flexibel rate not heavily tested! Warning, flexibel rate not heavily tested! Warning, flexibel rate not heavily tested! -- Started music on hold, class 'default', on Zap/4-1 Junk at the beginning 52494646 Skipped RIFF header! Warning, flexibel rate not heavily tested! -- Stopped music on hold on Zap/4-1 Warning, flexibel rate not heavily tested! Junk at the beginning 49443303 Warning, flexibel rate not heavily tested! Warning, flexibel rate not heavily tested! I'll throw a guess on this one. Sounds like the mp3 your playing is variable rate encoded (most mp3's are) and * doesn't like it. Convert the mp3 to nonvariable rate encoding, try like 128b, there's plenty of tools around to do it. I can't remember for sure, but the last time I tried a variable rate encoded mp3 it didn't work at all and had to convert it. That is if your mp3 is variable rate encoded. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone using * in a live production environment?
Ariel Batista wrote: Ok here is a short paragraph on our use of Asterisk in the real world. 1 inbound PRI ISDN 23/24 channel - Local phone service with 60 DID numbers. 1 Long Distance T1 line for inbound 800 numbers and all outbound long distance calls. running on P4 1.7G 512mg 20 gig HDD with 2 T400P boards. Ethernet on MB Intel 845G board. Hardware: 4 Adtran 750 with 24 FXS channels each. 1 Adtran 600 with 4 FX0 and 12 FXS ports. 1 ZetaFax server with US robotics modem. 1 HP Fax as backup 4 Inbound RAS lines for users 2 outbound RAS modems for dial out support lines. 40 452 phones (Really bad choice for phones) 10 390 phones (Again better then 452 but still bad phones) Cisco ATA 186 (nice works great) Cisco 7960 (Nice phone but worst phone to setup and maintain) 4 SIP phones X-Ten Lite with Telex USB connection to PC (Works great) Overall system is working with Support queues(AGI login user accounts) and meeting rooms. Voicemail system is not very good need some way to configure the boxes. They really need to redo this application for more standard settings. We have MOH working without any problems. Major down is no Graphical interface. No actual working manager. Got to get them to fix the Zombie lines. (I feel it's mainly do to our 390 and 452 phones. It works needs some fine tuning but it works. I have nothing good to say about the Aastra phones 390 or the 452. They are not really good for heavy use like we need! The Cisco 7960 is nice to look at but in the real world it's hard to get working and setup. If you don't know about Linux or are able to use scripts it's a real mess to keep up! This is where it's being held back as a real world player! This is the basic setup. Next step is outside offices connection. ___ What kind of resources are being consumed on the server? CPU,MEM,DISK,etc ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone using * in a live production environment?
Ariel Batista wrote: -- Original Message -- From: Ken Godee [EMAIL PROTECTED] Ariel Batista wrote: Ok here is a short paragraph on our use of Asterisk in the real world. ___ What kind of resources are being consumed on the server? CPU,MEM,DISK,etc I am not a Linux person (Trying to learn) so I am not able to check this out! But I do have over 12 gig of disk space still available. If you have some program or setting I can run on the server to give me this info I would love to see it! ___ Quick and dirty from console prompt you can use top From a desktop you can try ie.. xosview Or install something like gkrellm http://www.gkrellm.net ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Aastra 480 ADSI keypad problem
This is a really cool phone, except one problem, searched the archives and this was brought up before. Just wondering if anyone figured out how to solve it. I'm having the same problem as these previous posts... --- posted 06/09/03 Whenever I try using the voicemail through my ADSI display, it disables my # buttons. If I hit listen through the ADSI display, I can not delete messages. The 7 button no longer does anything... --- posted 06/09/03 I have the same problem. I use an Aastra 480 phone and as long as I don't touch any of the ADSI soft-buttons then my keypad stays active and the downloaded script works great. But as soon as I hit listen through the ADSI display, all of my normal 0-9*# keys get disabled and the script no longer maps any more options to my soft buttons. --- Any suggestions? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Aastra 480 ADSI keypad problem
Paul Crick wrote: It's more of an ADSI/Voicemail problem than phone specific I think? Or is it only affecting the 480s? I know I had a problem a while back with having the phone lock up and keypad become unresponsive, but with a newer version of Asterisk the problem went away. Since I'm only testing the 480, I'm not sure about other models. I talked with Aastra today and of coarse I got the answer I expected, We don't have anything to do with the ADSI programming, but the support (level 1) guy led me to believe that once you start using the any ADSI program that the keys on the phone become disabled? Well, that's what he told me anyway. I'm waiting for a Aastra senior engineer to get back to me on another issue with the phone, so I'm going to bend his ear and try to find out more. This is a really nice phone and would hate to disable the ADSI features because they don't work properly. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Clearing Queue Stats?
Is there a way to clear the Queue stats? That is with out restarting *? I'd like to reset them daily and don't see a way to do that. Unless the only way is maybe a cron restart asterisk like every weekday @ 04:00? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Live real extensions.conf samples?
It would be nice to see a real extensions.conf from a live business operation, every extensions.conf I've seen posted or been able to dig up so far would fail bad in a live business operation. I just have the beginings of mine and would like to make sure I don't miss anything. Most extensions.conf files I've seen wouldn't even let you dial 911 in thier dialplan. That's just something you don't want to forget! Not to mention that a business type extensions.conf needs to have several class of restrictions for different departments/people, most just have everything available to everyone, this is just not so in the real world. Not it mine anyway. If someone doesn't want to post you can alway email me direct. Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Which ADSI phones to buy?
Anton Tinchev wrote: Is there any verified source for unlocked aastra phones? Wade J. Weppler wrote: All of the Aaastra PowerTouch phones are ADSI capable (350, 390, 392, 480). You just have to make sure they are UNLOCKED or you have the security codes to be able to use the ADSI functions through Asterisk. There were some long discussions on the list a while back on this very issue. Best to search the list archives (google site:lists.digium.com). http://www.twacomm.com/Catalog/Jmp_Aastra/Product_brand.htm?SID=SGTJNV4JCA418NXJSKDT4VMHE4NL8XN8 After reading the list and seeing many problems with locked phones, when I was buying I asked lots of questions. If they don't know, go somewhere else. The vendor I bought my Aastra 480 from seemded to understand. One thing I noticed, when I received mine, is that on the side of the box along with the model code information it had Generic as part of the model code. I'm going to guess in that it means the phone has not been programed and is not locked. I'll only be buying the Generic models. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Answering Machine Detection
Why not just ask them to press-any-key ? And if any of them get confused you can refer them to Compaq frequently asked question #2859 http://web14.compaq.com/falco/detail.asp?FAQnum=FAQ2859 :) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X100P stopped working
I recompiled Asterisk with the aggressive echo cancellation on. That's all I changed, honest. After recompiling, it refused to run. I tried updating the source, etc, and eventually went back to no echo cancellation. Every time, I got this error while starting Asterisk. Please help! I have no idea what went wrong. Oh, and yes, wcfxo and zaptel are loaded, I checked with lsmod. I rebooted a few times too, to make sure everything had been cleared out. === [chan_zap.so] = (Zapata Telephony w/PRI) == Parsing '/etc/asterisk/zapata.conf': Found WARNING[1074404064]: File chan_zap.c, Line 6986 (load_module): Ignoring rxwink WARNING[1074404064]: File chan_zap.c, Line 626 (zt_open): Unable to specify channel 1: No such device or address ERROR[1074404064]: File chan_zap.c, Line 4949 (mkintf): Unable to open channel 1: No such device or address here = 0, tmp-channel = 0, channel = 1 ERROR[1074404064]: File chan_zap.c, Line 6730 (load_module): Unable to register channel '1' WARNING[1074404064]: File loader.c, Line 301 (ast_load_resource): chan_zap.so: load_module failed, returning -1 WARNING[1074404064]: File loader.c, Line 396 (load_modules): Loading module chan_zap.so failed! Warning, flexibel rate not heavily tested! Ouch ... error while writing audio data: : Broken pipe Just a thought... You did do a make clean first before recompiling? Couldn't tell from message if you just updated the source for asterisk or everything? The reason I ask because when I do updates I update everything ie. 1st, make clean,update Zaptel, make, make install 2nd, make clean, update Libpri, make, make install 3rd, make clean, update Asterisk, make, make install At least that's how I would do it, I believe asterisk relies on some shared libs when compiling and I want to make sure everything is matched up. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Context restrictions
One more question: What are agents, and what are they good for? Help and Wiki don't reveal much... I am starting to think we'd really need to get an overview of the * features and have that documented (without all the details, just to get the big picture which makes a start a lot (!) easier). Cheers, Philipp I'm a little new around here but.. From what I've been working on... Setting up agents in the agents.conf file allows you to then assign agents in your call queues as a members. Doing it this way allows a couple of things, like assigning agents to many queues and also most imporant, would allow agent to login from any extension. In a nut shell. agents.conf [agents] agent = 1001,4321,Ben Dover queues.conf [queue1] member = Agent/1001 extensions.conf exten = 28,1,AgentLogin(1001) exten = 29,1,Queue(queue1) Agent logs on, hears moh, waits for call. Inbound call gets transfered to x29 agents hears beep and inbound call gets connected to agent. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] managers.conf Clarification Question
Anthony Minessale wrote: Does Anyone have a breakdown on what each option means in manager.conf system,call,log,verbose,command,agent,user I want to make a user who does not get a ton of events in the socket and is just for sending a query and getting that 1 reply I dont want to keep restarting my pbx to figure it out. I'm sure some may be self-explanatory but I was wondering if anyone knows for sure which options are which. Anthony, I'm currently looking into the same thing, here's what I've been able to find out so far system = System events such as module load/unload call = Call event, such as state change, etc log = Log events verbose = Verbose messages command = Ability to read/set commands agent = Ability to read/set agent info user = Ability to read/set user info Also you do NOT need to restart or even reload asterisk to try the different settings, connecting/loging in, re-reads the manager.conf each time. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Adtran TA750 T100P
Jose Quinteiro wrote: Hello, So all the pieces are finally here, and I'm ready to play. I remember reading on this list that the connection Channel Bank - T100P requires a reverse cable. Is this a regular Ethernet reverse cable (i.e., only a couple of pairs reversed?) Please help me before I blow something up! Saludos, Jose. Here's a link to Adtran's site w/discrip for pin outs for loopback adapters and T1 crossover. http://www.adtran.com/adtranpx/Doc/0/BIAU1PH6DJBH39S2038BE81ID8/CU-94a6a9d76bfc11d78ff20c045003.html Here's a site that sells premade T1 cables by the foot. T1 cables should be 22awg solid, and each pair individually shielded. They also include shielded RJ connectors on there cables as well. $16.00 + .70 per foot, good cables. http://www.stonewallcable.com/product.asp?dept%5Fid=134pf%5Fid=SC%2D9598%2DX+++ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sound files
[EMAIL PROTECTED] wrote: I am still having trouble changing the sound files. I can take a wave file out of another program and set it in the folder and it will work If I record a wave file in Windoze No go Am I missing some thing ??? Thanks for the help Regards Mick sox file.wav -r 8000 -c1 menu1.wav Hope it helps. Ken ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] T100P to Adtran TA750 - No dialtone or ring
Jason Piterak wrote: Hello all, I've got a T100P connected to an Adtran TA750 with a T1 crossover... This connects to a patch panel with phone ports. The Adtran is fully populated with FXS cards. All I get on any phone port is a fast clicking noise... No dialtone. Asterisk 'sees' the card, (the channels show up in /proc/zaptel). Incoming calls are routed to the zap/x channel, but no ring. I'm hoping I'm overlooking something stupid. Thanks ahead of time... --Jason Does the light on front of the TA750 show that the channel is up? Can you do diags thru the TA750 admin port to your phones? ring, etc.? Do you have the most current TA750 firmware (L35?)? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] context confusion internal context 2 context only?
Andrew Joakimsen wrote: Includes are recursive Make a context with just all the internal extensions, and then make contexts for all the outbound calls and another group of contexts just as you are doing (admin, sales, etc) Thank you, Just the answer I was looking for! Ken ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] context confusion internal context 2 context only?
I'm trying to create several contexts for extentions with different levels of access to features and I'm wondering how the heck do I include all the contexts so that you can call internal to any extention in another context without giving the features of the higher level context to the lower level context? ie. [admin] include = local include = longdistance include = international include = services exten = 104,1,Dial(Zap/20|20) exten = 105 106 107.. etc [sales] include = local include = longdistance exten = 201,1,Dial(Zap/5|20) exten = 202 203 204 . etc [lobby] include = local exten = 303,1,Dial(Zap/10|20) exten = 304 305 ... etc Extentions in lobby should be able to directly call extentions in either admin or sales. Using an include in local then gives lobby access to ld,int.,services, etc. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] auto 'modprobe wct1xxp' on startup?
john lawler wrote: Hi guys, Thanks for your answers on my two questions yesterday. That's exactly what I was looking for, sorry for not noticing it myself, but I'm still getting acclimated to Asterisk and even Linux--from what I see so far, I love it. I've got another one now. Since my Asterisk install and configuration is fairly stable at this point, I'm interested it ensuring that during the event of a power failure, when the power returns (or if the machine is manually restarted) that Asterisk will successfully load on the other side (automatically). I've used the provided asterisk startup script (which I moved to /etc/rc.d/init.d) and RedHat's 'chkconfig' to make sure that Asterisk is started on bootup, but the problem I'm having has to do w/ the wct1xxp module, I believe. When I want to start Asterisk manually, I just type 'modprobe wct1xxp' and my two T1 cards are correctly started and then I can start asterisk w/ the normal commands and everything works. But, when I come back from a restart, it appears that the Asterisk startup failed, and I think it's b/c the wct1xxp module is not loaded. What is the recommended way to ensure this happens? I've been reading and found that modprobe (on startup, it appears) uses /etc/modules.conf, and here's what mine looks like: alias eth0 e1000 alias scsi_hostadapter megaraid alias usb-controller ehci-hcd alias usb-controller1 usb-uhci options torisa base=0xd alias char-major-196 torisa #post-install wcfxs /sbin/ztcfg #post-install wcfxsusb /sbin/ztcfg #post-install torisa /sbin/ztcfg #post-install tor2 /sbin/ztcfg #post-install wcfxo /sbin/ztcfg post-install wct1xxp /sbin/ztcfg #post-install wct4xxp /sbin/ztcfg (I commented out all of the modules I think I don't need, but it didn't work when they weren't commented out anyway). Does this have something to do w/ it? Do I need to add something to indicate that wct1xxp should be loaded on startup elsewhere? I appreciate your willingness to share your knowledge and expertise. jl This is the same problem I just had. Don't know if it's the best way, but it works. I created an executable file called rc.modules in my /etc/rc.d/ rc.modules- #!/bin/sh /sbin/modprobe wct4xxp --- and since the module needed to load before the init script called asterisk, I call the rc.modules file from within the rc.sysinit file (at the end of the file) /etc/rc.d/rc.modules boots with no problems now, otherwise asterisk would not just simply start by calling it from an init script. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE410P: Double/missed interrupt detected
Mark Spencer wrote: It won't cause any sort of serious problem, but you are getting it with an unusually high frequency. What's particularly interesting is that it occurs almost exactly once per minute, on the minute. This would seem to suggest that you have some hardware in your system which, once per minute, is blocking interrupts. Wasn't to hard to find. First thing I did was to unload the compaq adavance management driver. Now I just get a few on boot up, somewhere around when the kernels messing around with the mtrr serverworks chipset. Ok, I guess it's no big thing and shouldn't cause any harm but I've never had this type of error before on any Linux server I've run. I'll summize that it's one of two things.. a.) In an effort to push Linux's poor job of handling interrupt latancy, which makes it a not so good choice for real time systems that the TE410 driver is simply pushing the limits a little. Which is all right by me :) b.) The TE410 driver needs a little tweaking to play better with others and handle it's interrupt requests better. Which is all right by me too :) I'm not saying I know what the heck I'm talking about here but just curious. Falls into one of my Linux mottos... I now know more about it then I ever cared to Thanks, Ken ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] starting asterisk?
Ken Godee wrote: I'm trying to figure out how to start *. Rh7.3,CVS,TE410P,TA750 If I just try the way the docs spell it out /usr/sbin/asterisk -vvvc it fails.. Gezzz, must just be me but I'll get an email in the list incase anyone else can use in the future. Right or wrong, it's the only way I could get it going. Getting Asterisk to load and run on bootup was not as easy as just asterisk -vvvc, etc. * would fail just trying to start it without the TE410P module already loaded. Sooo.. Created an rc.modules file that modprobes wct4xxp and calling it from the end of rc.sysint. Then using a init script to start *. Tried to use the init script from gnuinter.net but had to change it to use asterisk instead of safe_asterisk or else it would fail. All seems good at this point. If anyone thinks the above is wrong/bad please... let me know, I just want to get going and have some fun working on configuring * :) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE410P: Double/missed interrupt detected
Rahul Arvind Jadhav wrote: hi, I have lately acquired a TE410P. The problem i face currently is that the span gets(or doesnt gets) UP in an uneven fashion i.e i have to load-unload the modules, wait for sometime and then start the application i dont know but this execise works well for me (though Rahul, You might want to check a couple of my previous posts in the last couple of days titled starting asterisk?. Don't know if this is the same type of trouble you're having, but I had a tuff time getting the whole app to fire up when booting, all working and booting well now. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] starting asterisk?
I'm trying to figure out how to start *. Rh7.3,CVS,TE410P,TA750 If I just try the way the docs spell it out /usr/sbin/asterisk -vvvc it fails.. /var/log/asterisk/messages Oct 3 22:23:34 WARNING[1024]: File chan_zap.c, Line 610 (zt_open): Unable to open '/dev/zap/channel': No such device Oct 3 22:23:34 ERROR[1024]: File chan_zap.c, Line 4930 (mkintf): Unable to open channel 21: No such device here = 0, tmp-channel = 0, channel = 21 Oct 3 22:23:34 ERROR[1024]: File chan_zap.c, Line 6711 (load_module): Unable to register channel '21-24' Oct 3 22:23:34 WARNING[1024]: File loader.c, Line 301 (ast_load_resource): chan_zap.so: load_module failed, returning -1 Oct 3 22:23:34 WARNING[1024]: File loader.c, Line 396 (load_modules): Loading module chan_zap.so failed! /var/log/messages Oct 3 22:03:12 cti-350 modprobe: modprobe: Can't locate module char-major-196 Oct 3 22:16:30 cti-350 last message repeated 3 times Oct 3 22:23:34 cti-350 last message repeated 6 times == If I modprobe -d wct4xxp and then start *, cb link comes up asterisk starts and all seems well. (well, except for the. kernel: TE410P: Double/missed interrupt detected) = Module slhc kname slhc objkey slhc names: slhc mode: NORMAL Module matching slhc: /lib/modules/2.4.20-20.7/kernel/drivers/net/slhc.o = = Module ppp_generic kname ppp_generic objkey ppp_generic names: ppp_generic mode: NORMAL Module matching ppp_generic: /lib/modules/2.4.20-20.7/kernel/drivers/net/ppp_generic.o = = Module zaptel kname zaptel objkey zaptel names: zaptel mode: NORMAL Module matching zaptel: /lib/modules/2.4.20-20.7/misc/zaptel.o = = Module wct4xxp kname wct4xxp objkey wct4xxp names: wct4xxp mode: NORMAL Module matching wct4xxp: /lib/modules/2.4.20-20.7/misc/wct4xxp.o = Zaptel Configuration == SPAN 1: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: FXO Kewlstart (Default) (Slaves: 01) ... Channel 24: FXS Kewlstart (Default) (Slaves: 24) 24 channels configured. /var/log/messages Oct 3 22:33:08 cti-350 kernel: TE410P: Span 1 configured for ESF/B8ZS Oct 3 22:33:08 cti-350 kernel: SPAN 1: Primary Sync Source Oct 3 22:33:08 cti-350 kernel: Registered tone zone 0 (United States / North America) *CLI Asterisk Ready. Ok, so I'm missing something? Combed thru docs, couldn't find anything. I take it that * autoload modules is not working in the order I need? So how can I get around it? Something in /etc/asterisk/modules.conf Do I just need to slip a modprobe wct4xxp in one of my rc or init.d files before firing up * ? What is the best way to start * on boot up? inittab or init.d? /etc/modules.conf options torisa base=0xd alias char-major-196 torisa post-install wcfxs /sbin/ztcfg post-install wcfxsusb /sbin/ztcfg post-install torisa /sbin/ztcfg post-install tor2 /sbin/ztcfg post-install wcfxo /sbin/ztcfg post-install wct1xxp /sbin/ztcfg post-install wct4xxp /sbin/ztcfg Thanks, Ken ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TE410P: Double/missed interrupt detected
Any ideas on the following? (CVS 10/01/2003) Only reference I could find was a Zaptel change log update... 2003-09-02 18:23 martinp * wct4xxp.c (1.6): Get rid of the Double missed interrupt message every time you load the driver and an email refering this to serial console usage. Something I should worry about? Oct 3 22:48:01 cti-350 kernel: TE410P: Double/missed interrupt detected Oct 3 22:49:01 cti-350 kernel: TE410P: Double/missed interrupt detected Oct 3 22:52:01 cti-350 last message repeated 2 times Oct 3 22:54:01 cti-350 last message repeated 3 times Oct 3 22:56:01 cti-350 last message repeated 2 times Oct 3 22:58:01 cti-350 last message repeated 2 times Oct 3 23:00:01 cti-350 last message repeated 2 times Oct 3 23:03:01 cti-350 last message repeated 2 times Oct 3 23:05:01 cti-350 last message repeated 2 times Oct 3 23:07:01 cti-350 last message repeated 2 times Oct 3 23:09:01 cti-350 last message repeated 2 times Oct 3 23:11:01 cti-350 last message repeated 2 times Oct 3 23:14:01 cti-350 last message repeated 2 times Oct 3 23:16:01 cti-350 last message repeated 2 times Oct 3 23:18:01 cti-350 last message repeated 2 times Oct 3 23:20:01 cti-350 last message repeated 2 times Oct 3 23:22:01 cti-350 last message repeated 2 times Oct 3 23:25:01 cti-350 last message repeated 2 times Oct 3 23:27:01 cti-350 last message repeated 3 times ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ADSI phones?
Any suggestions? The Aastra 480 390 seem popular along with the CybioLink. Does anyone use these phones (or others)? Are they compatible with atsterisk's ADSI? If so, how are people programming these phones? Searched thru archives, lots of previous talk but no soild info. I'd like to get a couple of ADSI phones to play with, just hate to waste the money if they won't work with *. TIA ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
