Re: [Asterisk-Users] Newbridge Mainstreet 3624 T1 channel bank now Alcatel
On Thu, 01 Jan 2004 14:54:45 -0500 Lance Arbuckle <[EMAIL PROTECTED]> wrote: > > > TC wrote: > > > > Hi > > I just came accross this > > Newbridge Mainstreet 3624 but the Alctel site appears to have zip for > > reference/user manuals > > Anyone by chance have 1 of these or a url for the docs ? > > Anyone know how to reset the passwords on the 3624 ? > > Also, is the 3624 suitable for use with the T100P and Asterisk. I was > considering one of these for my first asterisk channel bank since they > show up on Ebay regularly and fiarly cheap. > > -- > .~.Triad Internet Systems, Inc. > /V\Lance C. Arbuckle > // \\ 3315 Anderson Drive > /( )\ Winston-Salem, NC 27127 > ^'~'^ 336-771-2090 > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > I have one of these as well, its been sitting on a shelf till I get around to messing with it. I wasnt able to come up with any docs that would do me any good either. If anyone has had success with it and * I'd be interested to hear about it. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] analog or sip ? was far end disconnect supervision
On Sun, 11 Jan 2004 21:35:26 -0500 Lance Arbuckle <[EMAIL PROTECTED]> wrote: > > > Rich Adamson wrote: > > > > > Thanks to everyone that responded to my channel bank question. Ive > > > decided that the Adit 600 would be a good choice. > > > Then I got to thinking about SIP phones and wondered if their quality > > > has progressed to the point that they can be deployed to customers who > > > "just want their phones to work" and wouldn't tolerate any SIP hickups. > > > As for pricing, I would think the SIP phones would need to be in the > > > $200 price range to be competative with analog or ADSI phones plus a > > > channel bank. I know there are lots of variables that figure into the > > > analog vs SIP question like number of incoming lines and how they're > > > delivered and the number of extensions etc I guess what would be > > > helpfull to me would be some general rules of thumb that you asterisk > > > experts use to determine what type of extension phones to recommend for > > > a given customer. > > > > Lots of choices ranging from about $80 to $700 (and more) depending upon > > manufacturer, model, features, etc. Believe the wiki has some references > > to many of them. > > > > For business use, I've had excellent experience with the Cisco 7960 > > (refurb ~$350 with the power cube), and average-moving-to-good/excellent > > with the Snom 200 (using the latest firmware). Both are probably > > considered higher-end multiline business sip phones by most on this list. > > There are others but I've not attempted to eval those. > > > > Your customer is likely to drive the decision "if" you let them eval > > a few different models. Since you indicated that you're just getting > > started with *, etc, pure guess is that most business sales are likely > > to require a mixture of multi-line and single-line insturments. > > > > There has been a fair amount of list traffic relative to how various > > phones support nat, call transfer, music on hold, speaker volume, > > call waiting tones, and other issues. Best guess is that you would > > likely only sell a select set of single-line and multi-line units purely > > from a support perspective, with actual proposals based on specific > > requirements (eg, a location needs nat therefore this model, business > > office with all internal phones likely a different model, another > > business with an unlimited checkbook gets a Cisco ;). > > > > Rule of thumb... > > - don't give an executive or check-writer a cheap phone, or one that > >is so lite-weight they pull it around their desk > > - find a single-line instrument or two you are comfortable supporting > >(seems like the list has suggested at least one vendor's cheap phone > > has a high mortality rate that might be worth striking from your list) > > - understand where the ata-186 kind of boxes fit (and where they don't > >fit from a real business perpective) > > - understand the value (or lack thereof) for the phone having an > >internal switch with two RJ45 jacks (and who's phones don't work very > >well with this) > > - keep a sharp eye on the sip marketplace going forward ;) > > - understand the value of QoS in switches > > - find a supplier that delivers & invoices reliably, and will work with > >you on defective units > > > > If you're looking for opinions on specific models, I'm sure you'll get > > a number of responses from those with favorites. > > I haven't a clue, yet > > Correct me if I'm wrong, but it sounds like you're firmly in the SIP > camp. While I like the idea of adding extensions by simply plugging a > phone into the network, knowing what some of my potential Asterisk > customers have for data network hardware makes me cringe when I think > about adding 20,30 or 40 Sip phones to the mix :) > > So, I was thinking that perhaps going the analog route with a nice ADSI > screen phone might be best for those customers that are either (a)too > cheap to buy cisco's, (b)reluctant to replace network hardware, > (c)afraid of technology...etc. etc. From what I've pieced together from > googling the list archives it seems like this approach would offer the > customer a solid system today that could grow with SIP phones as > Asterisk and SIP mature a bit more. > > Can anyone share what their favorite analog business phones are ? > Is ADSI a good way to go ? If so, which models are your favorites? > > Thanks everyone :) > > -Lance > ___ > 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 > fwiw i just put in a 7x4 system at one of our smaller locations, i ended up using a used adtran ta-750(2fxo/4fxs), and at&t 972 analog phones. the main reason i used the at&t phones was the fact that they had the little 2.5mm headset jact like a cordless or ce
[Asterisk-Users] is it possible to turn auto answer off and on in the dialplan?
Is it possible to turn auto answer for the console off and on in the dialplan? If so would someone be so kind as to post a short example. I'd like to use the same sound card for external ringing over the paging system that I'm using for overhead paging. So my idea was to put the console in the group of phones to ring when a call comes in, which would ring over the speakers. But I'd like to keep being able to do over head paging by dialing an extension. I'd to have autoanswer=no in alsa.conf, and do something like ; overhead paging exten => 4600,1,SetMusicOnHold(silence) exten => 4600,2,SetAutoAnswerOff(CONSOLE/dsp) exten => 4600,3,Dial(CONSOLE/dsp) exten => 4600,4,Hangup instead of what i have now: ; overhead paging exten => 4600,1,SetMusicOnHold(silence) exten => 4600,2,Dial(CONSOLE/dsp) exten => 4600,3,Hangup Otherwise, I'd need to add another soundcard as console2 and run both outputs into the input of the paging system, correct? Thanks ahead, Jeff ___ 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] TDM card loses Dial tone
I have a cordless phone that causes this same thing to happen every time I plug it into a digium fxs port. I have an old style tdm card and a new one, same results with both. Don't know what it is about the phone that makes this happen; It works fine plugged into a pots line. The phone is a uniden 900mhz dss. -Jeff On Sat, 14 Feb 2004 14:49:49 -0500 Ulexus <[EMAIL PROTECTED]> wrote: > Same here. I, too have received replacement cards from Digium, and I have > even tried replacing the proSLICs, all to no avail. > > Also to note: the same port on each (of three) cards always goes out first. > > On Thursday, 12 February, 2004 19:22, John Vozza wrote: > > Same here... > > > > Usually after several of these show up in my system log: > > > > Power alarm on module 1, resetting! > > > > Need to unload/reload module wcfxs in order to get the dial tone back. > > Happens several times a week, sometimes more frequently. > > > > John > > - > > NetRom Internet Services973-208-1339 voice > > [EMAIL PROTECTED] 973-208-0942 fax > > http://www.netrom.com > > - > > > > On Thu, 12 Feb 2004, Youness El Andaloussi wrote: > > > I experienced similar problems too with a 4 chan tdm400. This seems to > > > especially happen when you make configuration changes. It has nothing to > > > do with runing X or no, it does not even have to do with redhat... I > > > experienced the same problem on mandrake. > > > > > > One thing you have to be extra careful is when restarting, make sure that > > > all the modules have entirely reloaded before expecting a dialtone with > > > an asterisk debug console "asterisk -r"... many of the times I > > > thought there was no dialtone and the asterisk process had gone cukoo, I > > > noticed that configuration was not entirely reload. > > > > > > Yet, reloading many times seems to get some of the TDM400 channels > > > hung. On the other hand, this problem does not seem to happen as > > > extensively when no reloads are made > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] what marks a vm message as old?
Hello, I've got a small asterisk system running in production. We use the voicemail mainly for letting people leave orders overnight and early morning in a special "order" mailbox. The manager at the site keeps complaining that the messages are always split into both old and new messages even though he deletes all the messages before he leaves and doesnt check them again until the next morning. I was just wondering exactly what all can happen that would mark a message as old? Do messages left the previous day automatically get moved to old? Thanks, Jeff Roberts ___ 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] what marks a vm message as old?
> On Thursday 04 March 2004 10:24, Jeff Roberts wrote: > > I was just wondering exactly what all can happen that would mark a > > message as old? Do messages left the previous day automatically > > get moved to old? > > Messages that have been listened to and are not explicitly saved to > the New folder are automatically moved to the Old folder when you > exit VoiceMailMain. > > So he would absolutely have to dial into that mailbox and check messages in order for some of them to be marked "old"? ___ 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: R: [Asterisk-Users] execute a context from cron
Michael George wrote: On Thu, Jul 01, 2004 at 03:58:25PM +0200, Manuel Wenger wrote: I want to have call forwarding (from the POTS) turned on at the close of work and turned off automatically by *. I would have a look at GotoIfTime: http://www.voip-info.org/wiki-Asterisk+cmd+GotoIfTime That should be much easier than a cron job It looks like this is just a conditional Goto and it will not spontaneously start a flow in a context. What I need is something that will, at a given time, act just like we picked up an internal extension and dialed a sequence of numbers. Thanks! Then you definitely want to take a look a sample.call, use the cron job to create you own file. ___ 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] odd behavior - adtran ta 850 + t100p
[EMAIL PROTECTED] wrote: I've been working with an adtran ta 850 hooked to a t100p pretty much all day today, and I haven't gotten past configuring zaptel.conf and zapata.conf. For some reason, when I pick up analog phone hooked up to the first module of a quad fxs card in the second slot of the ta 850, asterisk thinks that all four of the fxs modules in that card are going off hook. If I pick up a phone hooked to module 2 of the same fxs card then asterisk (correctly) only sees that module go off hook. When plugging a phone into any of the fxs pairs, I only get dial tone for a second or two and then I get silence. However, I can still dial extensions and get through. I'm not sure but maybe it is a config problem with the ta 850, as it takes a little more manual configuration than the ta 750 I worked with before. Anybody have any pointers? Here is the output on the console when I pick up a phone on module 1, and module 2, respectively: [EMAIL PROTECTED]:~# asterisk -r Asterisk CVS-HEAD-07/06/04-12:37:58, Copyright (C) 1999-2004 Digium. Written by Mark Spencer <[EMAIL PROTECTED]> = Connected to Asterisk CVS-HEAD-07/06/04-12:37:58 currently running on slack1 (pid = 702) - Remote UNIX connection -- Starting simple switch on 'Zap/5-1' -- Starting simple switch on 'Zap/6-1' -- Starting simple switch on 'Zap/7-1' -- Starting simple switch on 'Zap/8-1' -- Hungup 'Zap/5-1' -- Hungup 'Zap/6-1' -- Hungup 'Zap/7-1' -- Hungup 'Zap/8-1' -- Starting simple switch on 'Zap/5-1' Jul 6 22:30:03 WARNING[163850]: chan_zap.c:1288 zt_set_hook: zt hook failed: Device or resource busy Jul 6 22:30:03 WARNING[163850]: chan_zap.c:1288 zt_set_hook: zt hook failed: Device or resource busy -- Starting simple switch on 'Zap/6-1' -- Starting simple switch on 'Zap/7-1' Jul 6 22:30:03 WARNING[163850]: chan_zap.c:1288 zt_set_hook: zt hook failed: Device or resource busy -- Starting simple switch on 'Zap/8-1' -- Hungup 'Zap/5-1' -- Hungup 'Zap/6-1' -- Hungup 'Zap/7-1' -- Hungup 'Zap/8-1' -- Starting simple switch on 'Zap/6-1' -- Hungup 'Zap/6-1' Here is zaptel.conf: span=1,0,0,esf,b8zs loadzone = us defaultzone=us fxsks=1 fxoks=5-24 And here is zapata.conf: [channels] transfer=yes context=default language = en usecallerid = no hidecallerid = no echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 immediate=no signalling=fxs_ks echotraining=yes group = 0 channel => 1 context=trusted group = 1 signalling = fxo_ks rxwink = 300 channel => 5-24 Any help appreciated, -Jeff ___ 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 Well I tried setting up the the unused fxo ports, tried setting them to unused, and even moved the fxs cards around in the bank to see if it would make any difference. No joy though. Anybody know how to run some self tests on this bank to be sure its the problem? I'm pretty sure adtran will fix or replace the bank, but I'm sure they are going to want me to explain the problem but I'm not sure what info they'll need. ___ 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] Dell server for asterisk question!
I second that warning to stay away from the perc raid, I have one that continuously deals me fits. Leo Ann Boon wrote: The TE410P works with the 2650, I had 1 in there for months. One other thing, avoid the PERC RAID. The Linux driver in kernel 2.4 series is not very stable. FYI. Bartosz Jozwiak wrote: I am planning to buy Dell 2650 server with dual Xeon processors. And I would like to buy two TE410P cards for PCI with 3,3v. This is on Dell site about PCI slots for Dell 2650 server: 3 PCI-X (1x64-bit/133MHz, and 2x64-bit/100MHz) Does that mean I will be able to buy two TE410P cards ? Or I need to buy two TE405P cards ? Thanks for help. bartek ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Business Looking Analog Phone
Steve Totaro wrote: Thanks, they are kick ass. But this client wants a more traditional looking phone and they dont even want displays, lol, just a few speed dials. I guess I should have said that in the first post. - Original Message - From: "Andrew Kohlsmith" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Friday, May 14, 2004 9:08 AM Subject: Re: [Asterisk-Users] Business Looking Analog Phone I am looking for some analog phones are low cost yet not cheap looking. they should fit into a business setting. Can anyone help? Aastra PT390s. Kick ass phones. -A. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I went on this same quest earlier this year. AT&T and GE have some pretty good single and 2 line phones with and without displays, but they are really lightweight. officedepot.com has some of them, sometimes they have more to choose from in their stores. Staples also has a few. Aside from the weight, my biggest problem is somehow the users confuse asterisk into think them hanging up the phone is a flash-hook, but I think that is true for all analog phones. Panasonic also make a few nice phones but all I could find for purchase was white(which I didnt care for). All the phones I considered had the mini-jack headset plug too. ___ 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 7940 Phones as paging system?
Joe Antkowiak wrote: I am currently using 7960's with *, and line 6 is set to auto answer. Works great, customer is happy. As far as an intro-tone, you can set the dial command to play a sound (using the announce option) before the call is connected. I grabbed a simple tone wav file, and made it play that. Now, when the intercom ext is called, it plays the tone on the destination phone, and wa-la, intercom So it works. Let me know if you need sample configs. I'd like to see them, maybe you could put them on the wiki. -Jeff ___ 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 Partner Phones to SIP?
Matthew Branton wrote: I remember someone posting here some time ago about commercial offerings for taking channel banks of Avaya partner phones and turning them into asterisk compatible (SIP?) devices, but I can't seem to find a reference to the hardware manufacturer or specific experiences. Would anyone care to enlighten me? Off list is fine if this is a repeat, thanks very much. Matt http://www.citel.com ___ 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] tieline digit timeout
Tony wrote: I'm connecting to an NEC t1 card via t100p (working great so far!) however I'm having problems dialing from the NEC system to an asterisk extension (sip-grandstream). If I hit the trunck line and dial REAL quick 103 I get the sip extension ringing; if I don't I get an invalad selection message from asterisk - and I can see on the console only one or two digits arrived. How can I globally make asterisk wait for several seconds on that t100p? Thanks, t o n y ___ 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 Tony, I may not fully understand you question but it sounds like digittimeout is what you are looking for. http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+DigitTimeout Please keep me informed on your progress, off you list if you like. I'd really like to do pretty much the same thing towards the end of this year and I'm having a hard time making my vendor pbx vendor understand why/what I want to do. -Jeff ___ 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] Playing reveived message WAV file
Joseph wrote: What Linux players support WAV format? Many support wav. Few support GSM codec in wav format. You can use sox to convert to signed linear wav so you can play with all wav capable players. I think yours is the correct answer. The file is GSM codec in wav format. After converting it to MP3 I have no problem playing it. Is there a way to tell asterisk to send me MP3 format instead of WAV? Joseph, I had better results with mplayer than any other player I tried. Mplayer will play the .wav, and the .WAV files, which are the uncompressed and gsm encoded wavs. I could only get xmms to play the uncompressed wavs, whichever one that is. I tried several plugins and never got xmms to work with the gsm encoded wavs, although I could have been doing it wrong. I was never able to get mplayer to play the .gsm files. I tried vlc (videolan client) with some sucess with wavs but I ended up using mplayer. You might also try realplayer/helix player, I never got around to trying it. -jeff ___ 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] pickup any call
Andrew Kohlsmith wrote: I use it here all the time and it works very well. I have my home # ring a "dummy" line at work (it doesn't ring anywhere, just gets the call in to the office asterisk server) and then when my IM tells me I have an incoming call I can *8 it and receive calls to my number at work. :-) Hey Andrew, what kind of extension logic,etc to you use to get the im to tell you the call is there? ___ 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] paging/intercom
defiance wrote:
Hey guys, I have run into one last issue before I do my full *
conversion this evening. I can't seem to get paging to work. I have the
chan_oss module loaded as per the wiki, and I have the following in my
dial plan
;here is our intercom
exten => 6000,1,Dial,console/dsp
when I dial it here is the output from the console
-- Executing Dial("SIP/3062-4f07", "console/dsp") in new stack
<< Call placed to 'dsp' on console >>
<< Auto-answered >>
-- Called dsp
-- OSS/dsp answered SIP/3062-4f07
<< Hangup on console >>
== Spawn extension (from-sip, 6000, 1) exited non-zero on
'SIP/3062-4f07'
But I can't get anything to come out of the speakers. I know sound does
work on the system. Any Ideas?
Chris Locke
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Sounds like like your oss/alsa config doesnt like the drivers loaded.
If you have the oss module loading, try switching it to alsa or vice
versa. It can be a little picky, but if you have sound working for
another app, say aplay or play you should be able to get asterisk to use it.
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Re: [Asterisk-Users] Hardware for PBX with 4 incoming/outgoing lines and 20 phones
Andrew Elchuk wrote: Most of the time we are only using 3 of the lines for calling. If I used VOIP termination for outgoing calls wouldn't I still need the 4 lines to receive all of our incoming calls? Also do I need to have an IP phone to connect to a Wildcard TDM40B or could I just use the phones we have right now? Thanks. Don't know if your situation will lend itself to my solution, but it seems to work for me: I called and had the call waiting turned off on my pots line and set up busy call forwarding to my nufone number. Now, when someone dials my pots line, the first call comes in over a channel bank connected to a t100p. Each additional call, whether they call the pots line or the nufone line comes in over iax. I've only had a couple of simultaneous calls going, but I would think I could have as many as my internet connection will support bandwidth wise. I've been using it for a few weeks, and looking at the charges, its pretty cost effective to me. Unless your other three numbers are published or your faxing over several lines, I dont see why you can't shut them off and use one pots lines and voip for the rest. ___ 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] Playing GSM files
Brian wrote: Apple Quicktime will play gsm files iirc. Rodolfo Grave wrote: You can use WinAmp or xmms... it has a Plugin for playing GSM files.(not included in the standard installation but you can find it in google) RODOLFO Sys.Concept wrote: How to play GSM files? I want to go through some of them but I'm not sure which player to use. What do you guys use under linux to play gsm or wav49's? We have an environment that is all linux based terminals and I haven't found a player that will handle anything but the uncompressed wavs and work on the terminal using esd or nasd. ___ 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] Playing GSM files
Steven Critchfield wrote: On Mon, 2004-09-20 at 08:39, Jeff Roberts wrote: Brian wrote: Apple Quicktime will play gsm files iirc. Rodolfo Grave wrote: You can use WinAmp or xmms... it has a Plugin for playing GSM files.(not included in the standard installation but you can find it in google) RODOLFO Sys.Concept wrote: How to play GSM files? I want to go through some of them but I'm not sure which player to use. What do you guys use under linux to play gsm or wav49's? We have an environment that is all linux based terminals and I haven't found a player that will handle anything but the uncompressed wavs and work on the terminal using esd or nasd. sox gsmfile.gsm -sw -t wav - |esdcat What was wrong with your man(1) command? I should have been a little clearer. Its not me wanting to play the files, it would be the users, who don't have command line access. They also need the ff rw stop pause etc buttons. They'd be using the app to play voicemails out of their email, or vmail.cgi. It works fine off a windows box, but I have yet to come with anything that will play wav49 or gsm on a terminal with a gui. I did get mplayer to play a wav49 from the command line, but was never able to get the mplayer gui work on an ltsp terminal. ___ 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] system command vs mailfax and quotes?
hello,
been playing with spandsp, and rxfax. seems to work well.
i have not been able to run the mailfax command successfully from within asterisk.
[jeff-fax-in]
exten => s,1,SetVar(FAXFILE=/var/spool/asterisk-fax/${UNIQUEID}.tif)
exten => s,2,rxfax(${FAXFILE})
exten => h,1,system(/usr/bin/mailfax.sh ${FAXFILE} [EMAIL PROTECTED] ${CALLERIDNUM})
the console shows this command executing but i never get the email
however i can cut the command from the console, paste it to the command line and it works perfectly
such as:
/usr/bin/mailfax.sh /var/spool/asterisk-fax/1131570962.919.tif [EMAIL PROTECTED] 205550011
a little more testing shows me that at the command line:
/bin/sh -c /usr/bin/mailfax.sh /var/spool/asterisk-fax/1131570962.919.tif [EMAIL PROTECTED] 200011
fails though, which from what i read is how asterisk sends my arguments to the command line.
I've tried all sort of combinations of quotes and no quotes but can get anything to work.
anyone have an idea as to what im failing to do to get the arguments passed thru the system command correctly?
running CVS-v1-0-02/19/05-13:29:39
Thanks,
Jeff
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[asterisk-users] attended transfers going to wrong voicemail Asterisk 1.8 Polycom 650
Hi, I'm seeing an odd issue at a recent installation and have been unable to resolve it. Caller A calls Caller B and Caller B transfers to Caller C. Using a blind transfer, if Caller C doesn't answer then Caller A gets Caller C's voicemail. (as expected) However if doing an attended transfer (Caller B simply hits transfer, then transfer without announcing) then Caller A winds up in Caller B's voicemail box if Caller C doesn't answer. I realize my users are misusing "attended transfer" but it still doesn't seem to work as expected. I have tried setting both canreinvite=no and directmedia=no in sip.conf and it doesn't seem to make a difference. Asterisk is version 1.8.4.2. All phones are Polycom 650. I have another installation that is on Asterisk 1.6 with a mixture of cisco and polycom phones and I cannot reproduce the behavior there. Thanks in advance. If I need to provide more information please let me know. Thanks, Jeff Roberts -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
