Re: [Asterisk-Users] iax.cc opinion request
On 7/10/2005, "trixter" wrote: > I am considering using iax.cc (sixtel) and wondering if anyone had > opinions, good or bad. Are there outages with any regularity? How > responsive are tech support? How is packet loss? I am particularly > interested in termination to the UK, but will accept any comments > people have. Well - here we have a quandary. Opinion? Bad. (But so good) No outages that I can place on Sixtel - 24/7 rock solid - think a router hiccupped once for a couple of hours, but it wasn't theirs. Packet loss - again - as good or better that cell phones. Can't fault them (or him) there. UK termination (DID?) - can't say - thought they (or him) were US only. Tech support? Hahahahahahahahahahaha Ouch - my sides hurt! Took a month to get a DID. Still two months gone waiting for a Custom Toll Free - the primary reason I went with Sixtel... Though they take the money from your account as soon as you order. But then they don't continue billing until whatever you ordered works. It just takes SO long to get it working. Brett ___ 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] Systems Admin; Telecom Newbie - What do I need?
On 7/14/2005, "Ed Pastore" <[EMAIL PROTECTED]> wrote: >On Jul 14, 2005, at 11:54 AM, Adam Goryachev wrote: > >> Use a phone like the polycom IP301/501/600 which has a built-in 2 port >> 10/100 switch. ie, take the existing cable and plug it into the phone, >> then take a second cable, connect one end to the phone, and the >> other to >> your PC. No need for any additional major investment > >Independent of my telephony overhaul, I am planning on migrating my >network to gigabit to speed up some core file services (we do a lot >of server-based computing). Are there phones with a gigabit switch in >them? :) Ed, you should probably go with Cat 6 for your Gigabit network anyway. so just throw the 'cost' of the cabling on that 8-) Leave the 'old' 100 MBit plant in place and use that for VoIP. Then when you 'pull out' the old Cat 3 for your phones - you can use the Cat 3 for the 'snake' to pull the new Cat 6. Leave the VoIP network separate from the data network. Better for your 'QOS' too. Let's you add the 'perception' that the New VoIP will recycle the old equipment making it even a better 'deal'. Brett ___ 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] Phone manual..
On 7/15/2005, "Bill Wong" <[EMAIL PROTECTED]> wrote: >Hi, > >I tested asterisk server with Xpro program, and all the function working >well ( like 3 way calling, transfer ). But on the VOIP phone, I >don't know press which key for 3 way calling function and transfer >function... Can anybody teach me ? Might help to tell us the VoIP Phone you are using. Not all come capable of doing these things without some work. Brett ___ 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] Should this work?
On 7/25/2005, "Angus Comber" <[EMAIL PROTECTED]> wrote:
>I changed to:
>
>exten => _X.,1,Dial(ZAP/1/${EXTEN},60)
>exten => _X.,2,Hangup
>
>But still didn't work.
>
>even though could see channel with zap show channels - saw a channel 1
>
>Angus
Not as far as I know. ZAP channels don't get extensions (if it's TDM
card).
Just use exten=>_X.,1,Dial(ZAP/1,60)
But normally you use exten=>,1,Dial(ZAP/1,30)
where is the extenion you want to 'call' the phone line.
like I use one phone as so the dialplan know if I want to call
extension - it uses the exten=>,1,Dial(ZAP/1,30) and out she
goes.
Brett
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Re: [Asterisk-Users] Has Sixtel gone under?
On 8/2/2005, "Erik Espinoza" <[EMAIL PROTECTED]> wrote: >That's always been the site at that url. > >On 8/2/05, Tony Hoyle <[EMAIL PROTECTED]> wrote: >> Carlos Chavez wrote: >> > I have been using Sixtel from the beginning of the year and service >> > was >> > getting worse and worse. Yesterday I tried to access the website to get >> > the >> > CDR and I got an error saying that the domain no longer exists. I checked >> > the >> > whois and it says that the domain is on hold. Have they finally folded? >> > >> http://www.sixtel.net/voip/ doesn't look too promising... >> >> Tony I logged right on to the user login site OK - no problem. Well - other than all the trouble tickets I have with them have disappeared. Brett ___ 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] Compile ZAPTEL warning and Strange Congestion
Starting - oh - three weeks ago I started getting this when I compiled zaptel stuff: In file included from /lib/modules/2.4.26smp/build/include/linux/spinlock.h:6, from /lib/modules/2.4.26smp/build/include/linux/module.h:11, from wct4xxp.c:31: /lib/modules/2.4.26smp/build/include/asm/system.h: In function `__set_64bit_var': /lib/modules/2.4.26smp/build/include/asm/system.h:190: warning: dereferencing type-punned pointer will break strict-aliasing rules /lib/modules/2.4.26smp/build/include/asm/system.h:190: warning: dereferencing type-punned pointer will break strict-aliasing rules Just a warning - didn't bother me too much as everything worked any way. But it did come up in most of the source files. I think it's just a compiler option missing. 2.4.26 isn't exactly new 8-) I only use my * box for a home PBX - nothing critical about it. Then I started noticing this in the CLI: Jul 29 13:25:54 WARNING[9075]: chan_zap.c:3696 zt_handle_event: Ring/Off-hook in strange state 7 on channel 4 Jul 29 13:26:06 NOTICE[9078]: chan_zap.c:5797 ss_thread: Got event 2 (Ring/Answered)... Jul 29 13:26:12 WARNING[9078]: chan_zap.c:3696 zt_handle_event: Ring/Off-hook in strange state 7 on channel 4 Jul 29 13:26:25 NOTICE[9081]: chan_zap.c:5797 ss_thread: Got event 2 (Ring/Answered)... Jul 29 13:26:31 WARNING[9081]: chan_zap.c:3696 zt_handle_event: Ring/Off-hook in strange state 7 on channel 4 Jul 29 13:26:45 WARNING[9086]: chan_zap.c:5871 ss_thread: CallerID returned with error on channel 'Zap/4-1' I have this in my extension.conf [default] exten => s/6305551212,1,Congestion ; TEST OF CONGESTION It used to give a busy/congestion signal but now it just rings. "","6305551212","s","default","""ILLINOIS CALL"" <6305551212>","Zap/4-1","","Congestion","","2005-07-29 13:25:49",,"2005-07-29 13:25:57",8,0,"NO ANSWER","DOCUMENTATION" "","","s","default","","Zap/4-1","","Congestion","","2005-07-29 13:26:06",,"2005-07-29 13:26:14",8,0,"NO ANSWER","DOCUMENTATION" "","","s","default","","Zap/4-1","","Congestion","","2005-07-29 13:26:25",,"2005-07-29 13:26:33",8,0,"NO ANSWER","DOCUMENTATION" And it really looks like the incoming analog line is 'bouncing' or something. I don't think my zapata.conf has changed since I got the TDM411B. [trunkgroups] [channels] context=default switchtype=national signalling=fxo_ls rxwink=300 ; Atlas seems to use long (250ms) winks usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no context=default signalling=fxs_ks channel => 4 context=local signalling=fxo_ks callerid="Desk Phone" <6601> channel => 1 So anyway - I don't get it. Any one else have anything like this happening? Brett ___ 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] How to get in touch with sixTel?
On 5/20/2005, "Bryan Field-Elliot" <[EMAIL PROTECTED]> wrote: > If anybody here is a sixTel customer, can you share any tips & tricks > for getting in touch with anybody there? They are absurdly hard to get > a hold of, particularly when you have a technical issue needing to be > resolved. If anyone has any phone numbers other than their main 800 > line, I'd sure appreciate it. > > Thank you, > Bryan I'm in the same boat... I think we all are. People are there - they just don't WANT to talk to anyone. The beginning of this month the IVR changed - SOMEONE had to do that. The numbers I have are 800-987-9891 and 303-785-5014 and from the inetmedia llc web page I got 866-452-8930 and I thought once I heard the same IVR from all three but according to Google 866-452-8930 goes to some herbal-life place. LOL Could be! It didn't sound the same to me today. And then today - I saw my calls to them actually being charged against my sixtel account balance - something I hadn't seen before - so SOMEONE had to fix that. But from digging around the Internet I did find out they only started last August and it looks like a one-man-show so he may have been sick or in an accident since the end of last month (when I did get a trouble ticket resolved). Gee - that was about Noon on a Friday too! Hm... I am ready to contact the Denver BBB and Colorado State - I did find 'them' listed on the State business files. So I have 'their' addresses of record - I will wait and see if they bill me for another month for DID or Toll Free (both still 'Pending') and if they do - they get registered letters - and a call to the State and I'll pull the last payment back on the credit card just to pay for the letters. But their outbound sure works good for me... Brett ___ 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] RE: LOA for CFA . . work up "pencil copy"
ohhh! Two more companies to never deal with!!! LOL! Brett ___ 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] Qwest can't/won't
Comments at the end and in-line... On 1/18/2006, "Steve Murphy" <[EMAIL PROTECTED]> wrote: > Hello-- > > I've been making note of a local situation that seems somewhat > irritating, and thought maybe some of you experts out there, might be > able to propose some alternatives. > > The situation goes like this: > > Rural Wyoming. Cody, to be exact, but I'd wager that you could almost > anywhere in Wyoming and find similar situations. Or just about anywhere else QWest in involved... > The county gov. phones and the school system phones do not provide > callerid. > > Why? because they bought T1's at like $1k/mo instead of separate lines > at $100/line/mo each. Saves them a ton of money. Has been for years. > 5 or more years ago, callerid was not available out here. Slowly, over > the last 5 years or so, they have been upgrading the populace, so now, > almost everyone reports callerid, except the county gov., and the school > system. They come thru as "unavailable". They both want it, but they > can't afford to get/give it. They'd have to switch back to individual > lines to get that, at over 2x the cost, and they'd rather die. > > Now comes the interesting part. Qwest is offering most customers the > option to block anonymous calls. All you have to do is dial a *xx and > you won't get those pesky > charity/politicals/telemartket/pollsters/magazineresubscibers/etc. > > And you won't get the county gov. or the school system calls either. > > Why? Well, apparently, they say that the switches in Cody/Powell are > old, feature-free switches, and one is a slave switch to the other or > some such. And these old switches provide no capability to play with CID > on T1 lines. Probably still crossbar LOL > And they sure as heck are not going to upgrade the switch for a small > group of country bumpkins. What's interesting to me, as that of all the > features they send us little pamphlets about in our monthly bills, very > few of them are actually available here. Same with most other 'copper' providers - even if they have it - you can't get it without major bucks... > Are there any clever solutions to this problem, for the county gov or > school system? So - my thoughts (I used to live on the other side of the Bighorns) 1. What is the cost difference between a T1 'phone' and a T1 'data' line? One would think that the Wyoming government should 'help' or get a good rate on a data line... 2. Replace all your PBXs (like you have one LOL) with Asterisk 8-) 3. Get a VoIP provider (I use a TelIAX exchange out of Sheridan so my Mom can call locally) 4. Use you 'huge' user base to get a discount/special deal with the VoIP provider to get 'proper' CID AND a good rate (Hey helping out the gov and education is a feather for anyone - except QWest) 5. Drop all but 1 analog line for each location (to cover 911 calls) 6. Collect your $50 'bonus' for saving them thousands of dollars a month. and 7. Tell QWest to stick it Can't think of a better way to spend the summer (except camping) Brett ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Delay after first digit - dial plan
On 2/1/2006, "Tharindu Rukshan Bamunuarachchi"
<[EMAIL PROTECTED]> wrote:
>Dear Sir/Madma,
>
> I need to create dial plan to access out side world from office. Our
> office PBX system need to wait few seconds after pressing "9" before
> enter phone number.
>
> How should i prepare dial plan to add delay between first and rest of
> digits.
>
> Here is my idiotic try;
>
> exten => _9XX,1,Dial(Zap/1/9)
> exten => _9XX,2,Wait,2
> exten => _9XX,3,Dial(Zap/1/${EXTEN})
Tharindu -
Try this
exten => _9XX,1,Dial(Zap/1/9${EXTEN:1})
Basically - dial a the extension prepending a 9 and wait 4 * 0.5 seconds
but strip the first digit from the extension (you have manually dialed the
9 so it has to be stripped off).
You can play with the number of 'w' (a mini-wait command 8-) to get the
proper timing.
Brett
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RE: [Asterisk-Users] Voicemail Problem
Hack hack hack 8-) Now - comments inline...
> Here's the log of verbose level 3
>
> Asterisk*CLI>
> -- Playing 'vm-youhave' (language 'en')
> -- Playing 'vm-no' (language 'en')
> -- Playing 'vm-messages' (language 'en')
> -- Playing 'vm-opts' (language 'en')
> -- Playing 'vm-goodbye' (language 'en')
Here Asterisk says 'Goodbye'
> -- Executing Playback("SIP/210.23.1.139-081ee3d8",
>"Goodbye") in new stack
Oh! Looky Not Playing but Playback!!!
And it's looking for 'Goodbye' - not vm-goodbye not goodbye
> Feb 9 15:05:06 WARNING[23242]: file.c:509
>ast_openstream_full: File Goodbye does not exist in any format
> Feb 9 15:05:06 WARNING[23242]: file.c:821
>ast_streamfile: Unable to open Goodbye (format alaw): No such file or
>dire
> ctory
> Feb 9 15:05:06 WARNING[23242]: app_playback.c:132
>playback_exec: ast_streamfile failed on SIP/203.125.68.66-081ee3d8
> for Goodbye
> -- Executing Hangup("SIP/203.125.68.66-081ee3d8",
>"") in new stack
> == Spawn extension (default, 400, 3) exited non-zero
>on 'SIP/203.125.68.66-081ee3d8'
> Asterisk*CLI>
So apparently you have a 'h' extension and call 'Goodbye'
like:
exten => 'h',1,Playback(Goodbye);
which it ain't gonna find
Brett
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Re: [Asterisk-Users] setup default console debug levels
On 2/11/2006, "Oliver Rehak" <[EMAIL PROTECTED]> wrote: >Hi, > >how can i set-up that when i start the Console i dont have to type everytime >"set verbose 99" "sip debug" "oh323 debug toggle". In a config file or maybe >startup parameters? > >Thanx Last time I checked there is a lot of stuff you can put in 'asterisk.conf' Check the docs or READMEs or source - its not in the file. Brett ___ --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] How to test H.323
On 8/6/2005, "Frank Tarczynski" <[EMAIL PROTECTED]> wrote: > I'm trying to set-up H.323 support under Asterisk. I built a recent > CVS release and the ooh323c code from the asterisk-addons. Everything > built and installed and the H.323 stuff loads OK when asterisk starts. > > What is the easiest way to check if the H.323 code is working? I've > edited the h323.conf and extensions.conf files but I'm sure that things > aren't right. I've tried connecting to my asterisk box via netmeeting > but I'm having much success. I don't know if my conf files are > screwed-up or if ooh323c code isn't working. > > Can anyone help? You should have some 'new' CLI commands if ooH323 got loaded. h323 show config Show details on global configuration of H.323 channel driver h323 show peers Show defined H323 peers h323 show peer Show details on specific H323 peer h323 show users Show defined H323 users h323 show user Show details on specific H323 user h323 debug Enable H323 debugging h323 no debugDisable H323 debugging h323 conf is just about 'normal' except I did have to set the bindaddr to my machines IP - localhost or 0.0.0.0 did not work. [6633] type=friend context=default ip=192.168.10.22 ; UPDATE with appropriate ip address port=1720; UPDATE with appropriate port h323id=6633 disallow=all allow=ulaw ;allow=gsm outgoinglimit=2 incominglimit=2 ;e164=12345 rtptimeout=60 dtmfmode=rfc2833 I used a AT320 with the latest H323 firmware and it worked quite well. I have never used NetMeeting so can't help you with that setup. Brett ___ 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 call from iax extn but cannot call it - unableto cteate channel iax
On 8/7/2005, "Angus Comber" <[EMAIL PROTECTED]> wrote:
> Then in my extensions.conf I have:
>
> exten => 300,1,Dial(IAX/${EXTEN},20)
> exten => 300,2,Hangup
>
> I can dial from iaxComm (a soft IAX client) and that works fine. But
> when I try to dial 300 get:
>
> WARNING[22077]: channel.c1970 ast_request: No channel type registered
> for 'IAX'
> NOTICE[22077]: app_dial.c:777 dial_exec: Unable to create channel of
> type 'IAX'
>
> I have restarted Asterisk after config change.
>
> What have I not done. I am just testing the iaxComm program.
You have not used a correct Technology in your dial command.
The 'show application dial' says:
Dial(Technology/resource[&Technology2/resource2...][|timeout][|options][|URL])
Technology is the chan_.so file loaded and resource is the defined in
the configuration file for the technology.
So either you need a chan_iax.so - OR - you need to READ the
extension.conf
file. I have never seen a dial command like yours. One that is VERY
close
is one like Dial(IAX2/${EXTEN},20)
I am not just picking on you Angus. I do tend to read almost every
message
coming through the list and I get tired of reading all the questions that
5 minutes of reading the configuration files or searching the wiki (as out
of date as it is) or even typing 'help' at the CLI prompt can remedy.
Guess I'm just getting old...
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Re: [Asterisk-Users] Can call from iax extn but cannot call it -unableto cteate channel iax
On 8/7/2005, "Kevin P. Fleming" <[EMAIL PROTECTED]> wrote: >[EMAIL PROTECTED] wrote: > >> coming through the list and I get tired of reading all the questions that >> 5 minutes of reading the configuration files or searching the wiki (as >> out of date as it is) or even typing 'help' at the CLI prompt can remedy. > > And some of us are getting tired of people complaining about the wiki > being outdated/incorrect, when they have just as much ability to fix it > as anyone else does. At least I used a personal pronoun... or are you speaking for Digium? 8-) > It's especially annoying when someone posts a comment attached to a wiki > page saying "this is wrong here's the correct info' when they could have > just edited the page in the first place (which requires the same amount of > time). I don't do that - go find those that do if you're annoyed. > If you find a wiki page that is incorrect, incomplete or needs any > other editing, do it! The rest of the community will be thankful for > your help. Well - here on the list I have seen people still using 1.0.5. Maybe that info is correct for that version. I don't know. I'm not an expert. I tend to use the documentation supplied with and inside asterisk. And nothing you have said invalidates my statement. Most questions could easily be answered by reading... the config files, the help screens, or the wiki. I am sure the wiki will have massive updates after the 1.2 freeze hits. I am glad we have both had the chance to vent ___ 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 call from iax extn but cannot call it -unableto cteate channel iax
On 8/7/2005, "John Novack" <[EMAIL PROTECTED]> wrote: >[EMAIL PROTECTED] wrote: > > >> I am not just picking on you Angus. I do tend to read almost every >> message coming through the list and I get tired of reading all the >> questions that 5 minutes of reading the configuration files or >> searching the wiki (as out of date as it is) or even typing 'help' at >> the CLI prompt can remedy. >> >> Guess I'm just getting old... > > Guess we all are, but that is better than the other choice. > Many of us read all the posts, and some of us really get tired of a > small number who continue to complain and write paragraphs on those who > aren't able to either find the information they need or understand it > when they do. Well - the part snipped was the training part... 8-) > There is a large disparity between the beginner and those who have lived > with this camel ( Asterisk ) for months to years. > Help from the CLI leaves a LOT to be desired. The Wiki is either > correct, outdated, or wrong depending on which of the 1000 flavors of > Asterisk one happens to have settled on, usually because it mostly works > I could go on, but you get the idea. I think everyone gets that idea. > If you can help, do so, as someone else did for this fellow in just a > few characters > If all you can do is berate someone else for not reading and > UNDERSTANDING, or assume they didn't find the answer in the bushels of > poorly organized sometimes correct information, then just move on to the > next post. > > John Novack ( old fart ) Well - us old farts have to stand together (no one will stand near anyway). But I don't think just saying 'use IAX2' is going to enlighten anyone. That's why I included the whole Dial(Technology/resource) definition. There has to be a channel driver for the technology and the technology has to have a configuration file to assign the resources. I put that all in there so he could UNDERSTAND. And maybe others reading the list could understand it as well. And I guess IAX2 vs. IAX is a real bad example. LOL ___ 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 call from iax extn but cannot call it -unableto cteate channel iax
On 8/7/2005, "Zachary Whitley" <[EMAIL PROTECTED]> wrote: >> If you find a wiki page that is incorrect, incomplete or needs any >> other editing, do it! The rest of the community will be thankful for >> your help. > > I don't want to get in the middle of this but what wiki are we > referring to? voip-info.org/wiki-asterisk ?? I would be willing > to contribute if I knew were to go. Thanks. Yup - that's it. ___ 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] How to dial several extensions with differenttimeouts
On 8/10/2005, "Irakli Natsvlishvili" <[EMAIL PROTECTED]> wrote: > Hello, > > I know that using & it is possible to dial several channels. > > Question is - is it possible and if yes, how to dial several channels with > different ringing timeout? > > I mean the following - for example when SIP/500 is dialed, I want three > phones to be dialed simultaneously - 1000, 2000 and 3000. During 10 > seconds all phones are ringing, next 10 second phones 2000 and 3000 are > ringing and after 20 seconds only extension 3000 is ringing. > > If I use & in dial command, then all extensions are ringing >simultaneously, but ringing timeout after comma is set for all channels, > am I right? > > 500,1,Dial(SIP/1000&SIP/2000&SIP/3000,30) So be creative! 8-) 500,1,Dial(SIP/1000&SIP/2000&SIP/3000,10) 500,2,Dial(SIP/2000&SIP/3000,10) 500,3,Dial(SIP/3000,10) Total ring time 30 seconds after the call. This could get a little convoluted depending on what you really want to do with it. Brett ___ 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] AT-320 IAX & MWI?
I can't even get MWI to work under SIP! Course I haven't dug into it extensively yet... 8-) But your message did prompt me to check out www.aredfox.com and see if there were any updates. Wow - Version 1.44 just came out today (or was that yesterday over there?). Looks like they fixed a bunch of stuff. I have used this phone (as a phone) in all four protocols. Least testing was MGCP. Looks like I get to do it all over again. What kind of 'MWI' are you trying get? Stutter tone? Haven't got that yet. Brett On 8/26/2005, "Eric Lyons" <[EMAIL PROTECTED]> wrote: > Anyone out there using AT-320 phones with IAX? Can you get MWI to work? > I can only get it to work with SIP. > > Also, I note that (on the phone) dtmf has to be set to "sip info" or > dtmf doesn't seem to work at all (when the phone's under SIP). > > Sounds like the IAX firmware isn't all that finished? > > Eric. ___ --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
[Asterisk-Users] RE: Noise on ZAP channel
On 8/29/2005, "canuck15" <[EMAIL PROTECTED]> wrote: > > I have a couple SIP phones on a PIII 1Ghz 256MB * server with a TDM01B > connected to the PSTN. Calls between SIP phones are clear. Calls to > the PSTN are quite noisy. The other person does not hear noise but I > hear quite a bit. It is not an annoying sound but definitely much > noisier than typical PSTN or even cell phone calls. > > I believe I have a TDM400P REV H card. I definitely don't have any IRQ > issues. Everything not required is disabled in BIOS. Zaptel drivers > have been compiled with defaults and with MMX and other enhancements. > Have tried V 1.0.9.1 and current 1.2 beta1 software. Nothing changes. > Tried telco PSTN connection and VoIP provider connection via ATA which > both sound clear when connected directly to an analog phone. Nothing > changes. Tried adjusting RX/TX gain and echo cancellation in > zaptel.conf. Nothing changes. > > Does any one have any ideas? Could my FXO module be bad? Sure - it could be - I would check it in another TDM400 slot on the card to make sure. Also - an outside chance - make sure Tip and Ring are correct. You could be getting ground loops - depends on the noise. Brett ___ --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] RE: Noise on ZAP channel
On 8/30/2005, "Geoff Manning" <[EMAIL PROTECTED]> wrote: >[EMAIL PROTECTED] wrote: >> Also - an outside chance - make sure Tip and Ring >> are correct. You could be getting ground loops - depends on the noise. >> > > I am having noise and slip errors between my TE110P and a legacy PBX T1 > card. Could this be the same symptom? The connection is made using a 15 > pin serial on the T1 Card side to RJ48 on the TE110P side. I can't > determine what the pinout is on the serial side. Probably not Geoff. It is still digital at that point I think. It should be coming to you as a four wire balanced circuit. It depends on which legacy PBX you are using (tho it is pretty standard) And if it wasn't right - it probably wouldn't work at all. Brett ___ --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] Wierd Problem
On 8/30/2005, "Gulzar Hussain" <[EMAIL PROTECTED]> wrote: > Hi All > > I have posted this problem many times on the list but > no reply, trying one more time may be someone will > response this time > > When I call from 1 RTC Client to another without > Asterisk everything use to be fine but when asterisk > is there as a Registrar a problem use to occur in more > than 90% calls, Caller can hear the voice of the > receiving side but the receiver cant be able to hear the > caller for exactly 12 seconds, conversation will become > two way after 12 seconds. > > My Scenario > > Lucent Max TNT -> Asterisk -> RTC Client API > > Does anybody ever had this problem ? > Any suggestions will be higly apreciated > Thanx in Advance EXACTLY 12 seconds... Sounds like a timeout. Either the TNT or the Asterisk box is 'looking' for something - no idea which or what. DNS? CID? Something. Brett ___ --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] newbie install problem. And I alreadysearchedeverywhere!
On Sat, 2005-09-03 at 20:47 +0200, Jeroen Baten wrote:
> I just changed in zapata.conf the signalling to:
> signalling = bri_net_ptmp
>
> and this changes the output considerably:
>
> *CLI>
> -- Executing Dial("SIP/4001-2fea", "Zap/2/ stack
> -- Called -- Channel 0/2, span 1 got hangup
> -- Hungup 'Zap/2-1'
> == No one is available to answer at this time
>
> *CLI>
Ok - what the heck is this?v
Dial("SIP/4001-2fea", "Zap/2/http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [Asterisk-Users] Utility to find length of wav49 file
On 9/7/2005, "Malcolm Taylor" <[EMAIL PROTECTED]> wrote: >Thanks Flynn. > > Unfortunately the files aren't written by the voicemail application. > I was hoping that there was some little command-line utility which > would return basic sound information when passed the filename. > > Malcolm You could always use sox... I just ran it here $>sox -e stat Brett ___ --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] How to retrieve voicemail from an IP phone?
On 9/21/2005, "Min Qiu" <[EMAIL PROTECTED]> wrote: > Ok, I tried xlite (SIP softphone) and I could get into the > voicemail now. However, I got busy signal when I called > any Idefisk softphone from xlite. From Idefisk calling xlite > seems fine. > > Min Min - make sure your DTMF is working on the Idefisk softphone. You should be able to get to VM by calling the VM number, when it asks for Mailbox enter the number followed by the # and then the password followed by the # - it should immediately go to you mailbox - if it says Mailbox - waits 10 seconds and then says password and waits 10 more seconds - then your DTMF is not being recognized by the system. You have to fix that first. Otherwise - you have the dialplan set up to accept unregistered calls from the Idefisk phones. That's probably why you can't call them and why they don't get to the VM. If Idefisk is running sip they can make calls because they send a SIP setup with the number in it - but then can't pass DTMF. Brett ___ --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] Asterisk + GNUGK + Asterisk-Addons ooh323
On 9/23/2005, "Brian C. Fertig" <[EMAIL PROTECTED]> wrote: > I am having a slight issue. I am trying to register 2 asterisk boxes with > GNUGK and when I try to add the 2nd it gets denied cause of it saying its > a duplicate. How do I change the configs to allow more than one asterisk > box register to the same GK? > > brian Don't 'quote me' on this but... Look in the h323.conf/s and see if you have two different h323id strings for the servers. I think it defaults to 'Asterisk PBX' but the current sample makes it 'ObjSysAsterisk'. I am pretty sure they have to have different names or GNUGK is going to think they are the same. Brett ___ --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
[Asterisk-Users] PA1688 Phones using IAX MWI
Anybody have these working with Asterisk? I have an AT-320. Brett ___ --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] delayed pickup in ZAP interface and issue withhang up-s (fwd)
Pretty sure zapata.conf immediate=yes should do it for incomming calls and you won't have CID. Brett On 11/29/2005, "[EMAIL PROTECTED]" <[EMAIL PROTECTED]> wrote: >Hi out there, > >Im looking for support in regards to configuration of ZAP interface on a >TDM400P card (TDM04D).Im running Asterisk v 1.0.7 cvs on White Box 3.5 with >kernael 2.4.21 >Ive got two problems with Asterisk: >1. When there is an incoming phone call on the zap interface, Asterisk holds >call for about three rings and then forward it to a predefined extension. I >assuming that it waits for caller id, therefore Ive tried to switch it off >by modification of callerid and usecallerid with no positive outcome. > >My zapata.conf is: >[channels] >language=en >immediate=no >busydetect=yes ; to test when a line is hung-up >busycount=6 ; to prevent suprious hangups >context=from-pstn >signalling=fxs_ks >usecallerid=no >callerid= asreceived;(no or any other value doesnt bring any >improvement >to my setup) >echocancel=yes >echotraining=800 >callprogress=yes >group=0 >channel=>1-4 ___ --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: OH323 user configuration
On 12/6/2005, "Code Lover" <[EMAIL PROTECTED]> wrote: >Hi friends, > >Still i did not receive any instruction about my problem. I reached >somewhere but still my Asterisk not start to work as H.323 Gatekeeper. >I used the following configuration and i found that OH323 is >registered when asterisk starts. I'm pretty sure everywhere I see H323 mentioned - it specifically states the Asterisk is not and in all probability will never be a Gatekeeper. It is an 'end point'. Brett ___ --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] Recording Calls at the phone
On 1/6/2006, "Michael Sampson" <[EMAIL PROTECTED]> wrote: > Since not all of our operators are going through asterisk I can't switch > over to using asterisk. I agree that it is a much better system to > record the calls at the server, but thats just not an option. The call > recording software we use now is too integrated into our message taking > system not to use. Also the operators just make one 8 hour phone call > into our message taking system to get their "remote audio" so asterisk > would just record that as one long call, which won't work either. Anyone > have any other ideas. Michael - the clues are in what you originally wrote: >> Right now we use these recording controls from radio shack that plug >> in between the wall jack and the phone and plug in via a 1/8 inch >> stereo connector to the mic input on the computer. These phones have to be straight analog phones. Just put in a channel bank/TDM24XX/Sangoma whatever for the call center. Do not go IP phones there. Just wire them up they way they are now. If your investment in the call recording configuration is so great that you can't/won't change it - there is no reason not to give the rest of the company the benefits of VoIP. You just have to kick in some more money 8-) Since you say 'not all' are going thru asterisk - just put in a TDM4XX to test 4 agents or go with the ATA (re use them later for 'at home' users) once you have sold the idea to the 'powers that be'. Brett ___ --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] Register times out on internet outage
On 9/30/2005, "Chris Mason (Lists)" <[EMAIL PROTECTED]> wrote: > I am using AstLinux with > Asterisk CVS-HEAD-01/10/05-02:11:15-AstLinux built by [EMAIL PROTECTED] on a > i686 > running Linux > > On this box I am registered to two different providers for long distance > and international. If there is an internet outage of more than a few > minutes, I'm not sure how long it takes to make this happen, the > registration times out and when internet comes back the pbx never > re-registers and consequently Asterisk has to be reset. As there is > no-one there that can access the PBX it means the PBX has to be power > cycled, not an optimal solution. > Has anyone seen this problem, is it a CVS-HEAD problem? Chris - just do a reload. For my SIP stuff it comes back with a warning and I don't think it says anything for IAX. It times out after the 10th try (I think) and you should get notices in message file. Reload has always done it for me. Brett ___ --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] OOH323C
On 9/30/2005, "Dan Austin" <[EMAIL PROTECTED]> wrote: > Asking which H323 channel is the best turns out to be a deeply > personal issue, at least noting the responses in the past. You got that right! 8-) > I've tried and used all three. Here are my thoughts- > > Chan_h323 (the original)- > Did not work in our environment. Known issues with Cisco's > Call Manager. Other than the requirements for OpenH323 and > PWLib, it was easy to setup and configure. > > Chan_oh323 > Worked fine for us. Has the same dependencies as chan_h323, > also easy to setup and configure. > > Chan_h323 (ooh323c based) > This one has been a winner for us. No dependencies on OpenH323 > or PWLib, which while not terrible to build/setup, is extra effort > and can be tricky to match known working versions. > Setup and configuration has been very simple. If you have configured > the other channels, this one should seem familiar. > > A seperate note in favor of the new chan_h323 is the developer support. > I found a couple little bugs that related to our use of Cisco Call > Manager, and expected little or no interest in getting them resolved. > I had a test version made available to me in just over a day and > complete resolution a few hours later. Dan - as a thought - I am messing with a H323 'capable' IP Phone and I am (maybe foolishly) trying to use ooh323 with no gateway, gatekeeper, or anything else and I am not getting it to work too well. It seems 'sometimes' it does work. Is there any way - as far as you (or anyone else) knows that this will work with any flavor of H323 on Asterisk? I could just be messing up the configs. Brett ___ --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] strange wave like noise on sip handset
On Sat, 2005-10-01 at 14:47 +0100, Angus Comber wrote: > No it happens on our asterisk and at a customers. Not that noticeable but > not crystal clear. Didn't happen on a Snom 190. [Snip] > Sipura SPA-841 - when receiving an incoming call echoy for about 2-3 seconds > at start of call then echo went away. Remote end did not hear any echo. > Also wave like hiss as per my message. Angus - does the SPA-841 have AGC? Turn it off if you can. Brett ___ --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] spandsp and page orientation
On 10/5/2005, "Shawn Porter" <[EMAIL PROTECTED]> wrote: > samples are at > http://tumtum.no-ip.com/faxes/1128432831.3.tif > http://tumtum.no-ip.com/faxes/853107320051004-150908.tif > > Both of these were faxed from a Brother intellifax 750 through a ring-it > single-line simulator into my asterisk box (through an X100P clone) > both were normal 8.5X11 pages in portrait style (the map image should be > 8.5" wide and 11" long) > > I can't take the old fax machine offline until I get this resolved. If > anyone has any ideas I am open to suggestion. Actually Shawn - those are 'beautiful' faxes... They are done in 'fine' mode which doubles the horizontal DPI. I forget the conversion method but I know when I was using EFax and HylaFax they would both do this on a 'fine' fax. All you have to do is find the right 'post-processor' to get these to print right. Brett ___ --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: RES: [Asterisk-Users] CDR MySQL
There is a recent fix to cdr_addon_mysql.c that added 5 retries and a timeout variable to the conf file. You might want to try that. Brett On 10/6/2005, "Jozeph Brasil" <[EMAIL PROTECTED]> wrote: >I don´t have access to MySQL administration! :( >I´m using a remote database to store... > >-Mensagem original- >De: Steve Davies [mailto:[EMAIL PROTECTED] >Enviada em: quinta-feira, 6 de outubro de 2005 09:48 >Para: Jozeph Brasil >Assunto: Re: [Asterisk-Users] CDR MySQL > >The usual "fix" is to reconfigure MySQL - It is the database that is >causing the timeouts. I do not know how this is done. Perhaps Google >for "MySQL connection timeout" and go from there? > >Regards, >Steve > >On 10/6/05, Jozeph Brasil <[EMAIL PROTECTED]> wrote: >> Hi Steve, >> >> Thank you for reply! :) Do you know if anyone have been patched to solve >this? >> >> Quoting Steve Davies <[EMAIL PROTECTED]>: >> >> > I do not know for sure, but in my experience, the most common cause of >> > this error is an application which assumes a database connection will >> > stay open forever. >> > >> > Some MySQL installs have a timeout on idle connections. It should be >> > possible to set this to a very-long or infinite timeout. >> > >> > Hope that helps, >> > Steve >> > > > >___ >--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 > ___ --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: RES: [Asterisk-Users] CDR MySQL
You 'should' be able to go into /usr/src/asterisk-addons/ and do a 'make update'. This should only update the addons directory... But I have been wrong before... Brett On 10/6/2005, "Rosario Pingaro" <[EMAIL PROTECTED]> wrote: >where can i found it? >thanks ___ --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] Asterisk and firewall
On 10/7/2005, "Arjan van Eersel" <[EMAIL PROTECTED]> wrote: > Hi all, > > I have installed an asterisk server at my office, the server is behind a > firewall. On the firewall I've set NAT a rule for incoming traffic on > port 5060 to be forwarded to the server. > > Connecting from home with my sip client doesn't work at all. > > The asterisk server itself is ok, when I make a local connection at my > office, 10.0.0.129 (client) to 10.0.0.6 (asterisk server) it works all > perfect. > > Should I perhaps open more ports in the NAT settings? Oh ya - this is SIP! Check you rtp.conf and forward those ports UDP as well. And set your SIP client to use them if you can. Otherwise change them. Brett ___ --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] Asterisk and firewall
On 10/7/2005, "[EMAIL PROTECTED]" <[EMAIL PROTECTED]> wrote: >On 10/7/2005, "Arjan van Eersel" <[EMAIL PROTECTED]> wrote: > >> Hi all, >> >> I have installed an asterisk server at my office, the server is behind a >> firewall. On the firewall I've set NAT a rule for incoming traffic on >> port 5060 to be forwarded to the server. >> >> Connecting from home with my sip client doesn't work at all. >> >> The asterisk server itself is ok, when I make a local connection at my >> office, 10.0.0.129 (client) to 10.0.0.6 (asterisk server) it works all >> perfect. >> >> Should I perhaps open more ports in the NAT settings? > >Oh ya - this is SIP! Check you rtp.conf and forward those ports UDP as >well. > >And set your SIP client to use them if you can. Otherwise change them. > >Brett AND read the wiki. Brett ___ --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] Busy not jumping n + 101 anymore
On 10/14/2005, "William M. Sandiford" <[EMAIL PROTECTED]> wrote: > I recently upgraded my Asterisk system to the latest CVS-HEAD > > Asterisk CVS HEAD built by [EMAIL PROTECTED] on a i686 running Linux on > 2005-10-12 13:34:09 UTC > > Ever since this upgrade, the system is jumping n+101 if it gets a busy > on a Dial command, it is now proceeding to the next priority (n+1) > > Has something changed with this? Is there a way to change it back? So glad to see you read the documentation... Try scanning UPGRADE.txt A lot has changed. Brett ___ --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] Busy not jumping n + 101 anymore
On 10/14/2005, "Jeremy Gault" <[EMAIL PROTECTED]> wrote:
>[EMAIL PROTECTED] wrote:
>
>>On 10/14/2005, "William M. Sandiford" <[EMAIL PROTECTED]>
>>wrote:
>>
>>
>>
>>>Ever since this upgrade, the system is jumping n+101 if it gets a busy
>>>on a Dial command, it is now proceeding to the next priority (n+1)
>>>
>>>Has something changed with this? Is there a way to change it back?
>>>
>>>
>>
>>So glad to see you read the documentation...
>>
>>Try scanning UPGRADE.txt
>>
>>A lot has changed.
>>
>>
>We've had the same problem here ever since we upgraded to CVS-HEAD.
>When someone placed a call to a number that was busy, they would just
>receive the "call cannot be completed" recording we have setup at n+1.
>
>Not to sound nitpicky or hateful, but I just reviewed UPGRADE.txt again
>here and I don't see anything about it. If it is in there, could you
>please point it out to me? (Seriously, as I didn't see it.) If it
>isn't, someone with CVS access should probably add it in.
>
>Now, I will say that I'm assuming (from the new behavior and the "show
>application dial" output) that one should now be using the ${DIALSTATUS}
>variable to handle these conditions. (i.e. from your dial, make n+1 be a
>Goto(s-${DIALSTATUS}) command, and create s-BUSY, s-CONGESTION, etc. in
>the same context.) Once I get around to updating our dialplans, that's
>what I plan on doing.
>
>Someone please correct me if I am wrong. *dons asbestos armor, just in
>case*
No armor required...
I was wr-wr-wr-wrong... It is in extentions.conf. As mentioned by
others.
Well - I been wrong before... 8-)
Brett
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Re: [Asterisk-Users] Attempted to delete nonexistent schedule entry...
On 10/15/2005, "J. Iddings" <[EMAIL PROTECTED]> wrote: >I'm also having this issue. Everything seems to work, but it's an >unnerving error. Any thoughts? > >Jimmy wrote: >> I just upgraded my test Asterisk box to the latest CVS HEAD. "show >> version" only shows "Asterisk CVS HEAD built by rootetc", with no >> date or version number. I downloaded this version on Monday, Oct 3. >> About once every minute, I get this while at the CLI> prompt: >> >> sched.c:296 ast_sched_del: Attempted to delete nonexistent schedule >> entry 1! >> >> This only appeared after updating. All functions seem normal, other >> than these messages. Phones work, auto-attendant works, voicemail works, >> etc. What's going on? OK - I been wrong so many times this week - it ain't funny... But - I think - this part of the scheduling change to the registration stuff. In one update, when a remote phone/system stopped responding to qualify attempts, the system would stop trying to verify the connection. Forever. Not exactly a 'good thing'. It would tell you that by saying Forever but still not good. Then an update added some stuff to ?iax.conf? like: ;qualify=yes ;qualifysmoothing = yes ;qualifyfreqok = 12 ;qualifyfreqnotok = 3 to modify how and when the system would retry these connections. During the time between the first and second update, I would get these messages when I did an iax2 reload. It had stopped trying to qualify the connection - and then the reload would start it backup. It would 'inform' me with the 'attempted to delete nonexistant schedule entry' because the time of the next scheduled event was no longer active. So in essence - it is a warning and not an error. Brett ___ --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] Busy not jumping n + 101 anymore
On 10/15/2005, "John Novack" <[EMAIL PROTECTED]> wrote: >Eric "ManxPower" Wieling wrote: > >> Andrew Kohlsmith wrote: >> >>> On Friday 14 October 2005 20:50, Eric "ManxPower" Wieling wrote: >>> >>>> Oddly enough, I believe it's mentioned in UPGRADE.txt. >>> >>> Care to tell us where? I just checked my CVS HEAD copy of UPGRADE.txt. >> >> >> Sorry, it's in asterisk/configs/extensions.conf.sample > > Which isn't even produced if one doesn't "make samples" > > What backwards thinking put the information there, and in addition > changed the way jumps used to work as the default? > If more time were spent on fixing things that were broken, and making > the interface to the existing PSTN analog lines work smoother there > might be more acceptance. > > JMO Well - number 1 - it IS - CVS HEAD. Next - I always run "make samples". In the /etc/asterisk/ directory it renames all your old config files that have changed to *.old. So long as you don't stop and restart Asterisk - its fine. Just diff the two files and you get to see the differences. Now that seems a 'funny' thing to do as you are thinking - heck ALL my configs are 'different' - as are mine. But I keep a subdirectory full of MY config files (just in case) and I also keep a subdirectory full of the LAST updates config files just to compare against. After seeing what's new - fixing MY config files - moving the NEW ones to the latest subdirectory - moving MY config files back to the working directory - THEN I restart Asterisk. It's called system administration. One day I'll set it up with a script, but right now it isn't bad or too much work. But I do update almost everyday. Brett ___ --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] No Audio from Console but mpg123 fromshellworksfine.
On 10/16/2005, "Jonathan k. Creasy" <[EMAIL PROTECTED]> wrote: >-Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir > Cohen > Sent: Sunday, October 16, 2005 2:59 AM > To: asterisk-users@lists.digium.com > Subject: Re: [Asterisk-Users] No Audio from Console but mpg123 from > shellworksfine. > >> Do you use ALSA or OSS for sound? What kernel version? > > ALSA. I used alsactl to reset the mixer controls as it was muted by > default. I'm running CentOS 4.1, I don't remember the kernel version > right off and I don't have access to that box here, I'll check it from > work tomorrow. > >>> [chan_oss.so] => (OSS Console Channel Driver) >>> == Parsing '/etc/asterisk/oss.conf': Found >>> == Registered channel type 'Console' (OSS Console Channel Driver) > >> Asterisk grabs /dev/dsp . I figure you can't play anything at this >> point. Though you should get stuck at trying to open it. Sigh... >From modules.conf ; ; Load either OSS or ALSA, not both ; By default, load OSS only (automatically) and do not load ALSA ; noload => chan_alsa.so ;noload => chan_oss.so Bet the problem is around here. Brett ___ --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] Cannot telnet to port 5038 on asterisk
On 10/17/2005, "Chuck Bunn" <[EMAIL PROTECTED]> wrote: > Hi, > > I cannot do the following: > > telnet 127.0.0.1 5038 Is telnet enabled? Brett ___ --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] Need language variable to user account
On 10/17/2005, "Ronald Wiplinger" <[EMAIL PROTECTED]> wrote: > My users do have different language requests. I would like to give them > their wish language. > > I could setup an extra database for that. > I wonder if it would be much work to add this field in sip.conf (and > realtime)? Ronald... IF I had customers who needed different languages via sip.conf... I would use the - er... language= setting in there. But I don't know if realtime uses it - and I don't know what version of asterisk (or [EMAIL PROTECTED] - or whatever) you are using. Brett ___ --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] Cannot telnet to port 5038 on asterisk
On 10/17/2005, "Michael Furdyk" <[EMAIL PROTECTED]> wrote: > He is just using telnet to check for the port being open/working... (not > telneting to the telnet port) > >-- Mike > >-Original Message- > [EMAIL PROTECTED] >Sent: Monday, October 17, 2005 12:28 AM >Subject: Re: [Asterisk-Users] Cannot telnet to port 5038 on asterisk > >On 10/17/2005, "Chuck Bunn" <[EMAIL PROTECTED]> wrote: >> Hi, >> >> I cannot do the following: >> >> telnet 127.0.0.1 5038 > >Is telnet enabled? > >Brett Here it is Sunday - And I been wrong already this week... Is manager.conf 'enabled=yes'? Brett ___ --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] Need language variable to user account
On 10/17/2005, "Ronald Wiplinger" <[EMAIL PROTECTED]> wrote: > [EMAIL PROTECTED] wrote: > >> On 10/17/2005, "Ronald Wiplinger" <[EMAIL PROTECTED]> wrote: >> >> >>> My users do have different language requests. I would like to give them >>> their wish language. >>> >>> I could setup an extra database for that. >>> I wonder if it would be much work to add this field in sip.conf (and >>> realtime)? >>> >>> >> >> Ronald... >> >> IF I had customers who needed different languages via sip.conf... >> I would use the - er... language= setting in there. >> >> But I don't know if realtime uses it - and I don't know what version of >> asterisk (or [EMAIL PROTECTED] - or whatever) you are using. >> >> Brett > >Brett, > > how would you do that? Giving each language group a different context? > At least that was I came up with. > Actually the question is going even further, Think about that: > > I will create 20 features, 20 different pay plans (tariffs), > One of the 20 features is the language. > If I would use context, I have soon 400 x 10 possible languages > (slightly exaggerated, hehehehe) > > I guess, if I know how to add languge, than I can add the other features > as well, ... > > My next try is to setup a feature mysql database for each user. This > database will be queried at the beginning of the context and than give > you all the variables you may need, ... > Maybe something like that exists? Apparently I misunderstood the use of the word 'language'... You mean English, Spanish, Italian, and German - or the actual wording of prompt itself? For a language - set the 'language=en' or 'language=es' in the sip.conf for that user. It is 'supposed' to be carried through. Should be something on the wiki about it. If you mean the wording or the prompts/IVR etc - well - that's why you get the 'big bucks'. 8-) Brett ___ --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] Cannot telnet to port 5038 on asterisk
On 10/17/2005, "Chuck Bunn" <[EMAIL PROTECTED]> wrote: > Hi, > > Yes it is enabled I have even checked various logs and nothing... I > checked '/var/log/messages', '/var/log/secure', > '/var/log/asterisk/full', and even '/var/log/mysqld.log' nothing, nada, > nein - its odd that a failed connection attempt is not logged somewhere, > perhaps I must somehow turn logging on for the asterisk management > portal. Any ideas? Are you 'sure' Asterisk is running? ps ax asterisk -r (which maybe shouldn't work if you can't telnet...) asterisk -c ? ends with a CLI> ? And stays that way? Brett P.S. I just connected and it didn't even show that THAT occurred... ___ --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] Delayed ringing on some SIP phones
On 10/17/2005, "Chris Bagnall" <[EMAIL PROTECTED]> wrote: > Hello all, > > One of the buildings I have an asterisk box deployed in is used by two > small companies on two floors. They have an agreement between them > whereby they'll answer each other's incoming calls and take messages if > the office is empty/everyone is on the phone. > > Each of them has an ISDN BRI delivered to asterisk via zaphfc, then > dropped into a context as follows: > exten => s,1,SetCIDName(Company 1) > exten => s,2,Dial(SIP/200&SIP/201&etc.,30) > exten => s,3,Voicemail(su200) > > Each company is able to see on the LCD on their SIP phones whether the > call is for them or the folks up/downstairs. > > What I'd like to do is implement a delayed ringing strategy - i.e. if the > call comes in for Company 1, only their SIP phones will ring for the first > 15 seconds, then if there's not been an answer, company 2's SIP phones > will also start ringing. > > Is there any way to do this without stopping Company 1's phones ringing > (i.e. timing out the dial statement after 15 seconds)? Bingo! You got it! Timeout the dial after X seconds - and then do a Dial to both companies for another another X seconds. Remember - busy does a jump to n+101 (some one is there...) and unavailable just goes to the next step. Brett ___ --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] Extensions
For analog phones - same thing 8-) except it is in zapata.conf mailbox=whatever ; under (er just above) the channel. Should give you a stutter dial tone. Brett On 11/2/2005, "Adam Moffett" <[EMAIL PROTECTED]> wrote: > >This works for me, your mileage may vary: > >in sip.conf add these two lines under the sip user: >mailbox=/[EMAIL PROTECTED] >/notifymimetype=application/simple-message-summary > >example for mailbox= >[EMAIL PROTECTED] > >The SIP device must support this feature of course. >And if you're not using SIP then I have no idea what to do :) > > > >Andrea Frigo wrote: > >> How can I configure Asterisk to tell me if there are messages on my >> voice mail as soon as I hook up an internal phone? >> >> Regards, >> Andrea Frigo ___ --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] Timestamps in Console?
There's a whole new section called [options] Normal 'commandline' options go in there. Read the source 8-) Brett On 11/3/2005, "[EMAIL PROTECTED]" <[EMAIL PROTECTED]> wrote: >[EMAIL PROTECTED] wrote on 11/03/2005 11:53:17 AM: > >> Chris Wade wrote: >> > Use 'timestamp=yes' in asterisk.conf instead of -T. >> > >> > -T only affects messages generated by THIS connection (ie asterisk -RT > >> > generated messages... not server generated messages. >> > >> > 'timestamp=yes' affects all messages generated. >> > >> >> And after adding timestamp=yes to asterisk.conf, don't forget to restart > >> asterisk (not just the console) to make the change take effect. > >Where should it be added? Mine is a default setup. It has two sections: >[directories] and [files]. > >Tim Massey > > ___ --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] SIP Disconnect Supervision
voicemail.conf ; Minimum length of a voicemail message in seconds for the message to be ; kept The default is no minimum. minmessage=3 Does it for all channels 8-) Brett On 11/3/2005, "Steve Blair" <[EMAIL PROTECTED]> wrote: > > I have a case where a call from the PSTN to our SER proxy goes unanswered. >As a result it is relayed to our Asterisk server for voicemail. However >before >the greeting plays the caller hangs up. This results in an empty message >being created and emailed or the mwi gets activated. > > I saw a few posts about Disconnect Supervision and Disconnect Supervision >with inbound SIP connections but I did not see any resolution to the SIP >question. > > Does this sound like a SIP Disconnect Supervision issue? If not what seems >to be the issue? Also does anyone have any suggestions on how to stop >these messages from being created? > >Thanks,Steve > >-- > >ISC Network Engineering >The University of Pennsylvania >3401 Walnut Street, Suite 221A >Philadelphia, PA 19104 > > >voice: 215-573-8396 > > 215-746-8001 > >fax: 215-898-9348 > >sip:[EMAIL PROTECTED] > >___ >--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 > ___ --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] use AT320 international call
On 6/12/2006, "Min Qiu" <[EMAIL PROTECTED]> wrote: >Hi all, > >The firmware I used is pa168s_iax2_us_151011.bin. > >My problem is the handset dial before I finished key in all >the numbers, no matter how fast I managed to press the keys. >It appeared it always dialed immediately, for example "011862", >when I actually ment to dial 0118620. Thus left the >remaining numbers "0" unsent. > >The handset had its dial plan disabled. It configured to use >iax protocol. My extensions.conf has this: > >exten=_01186.,1, dial(SIP/,60) > >and it works fine with other iaxy and Cisco ATA. > >Anyone encounter this symptom? Can you share your experience? > >Thanks, > >Min Min _ I amybe shooting in the dark here but - are you using inband DTMF? My AT320 only dials after I hit the Dial button or it times out. I do NOT use inband DTMF. Brett ___ --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] Stuck. Extenions.conf? Realtime? MySQL? Grrrrr!
On 3/15/2006, "Douglas Garstang" <[EMAIL PROTECTED]> wrote: >Boy, am I stuck... [snip] >My brain hurts. Doug, Whenever I have gotten to this point in a project, I use two rules for handling the situation. Rule 1. Booze Rule 2. Throw money at it. Rule 1 makes me feel better. Rule 2 takes care of the problem but... If the boss isn't happy - fall back to Rule 1. The hardest target to hit in the programming shooting gallery is the moving one. Unless you 'sold' the powers that be that Asterisk is the answer to all questions... then you made your bed... but as I remember, I think you got 'stuck' with this one. You can probably (but I doubt it) buy a system that will do all this for you. Probably not out of the box tho and probably not without a large 'programmers' bill to boot. And several third-party packages. So grab your favorite alcoholic beverage, nail down what they want, and start solving the problems. Even if it takes a year - it will be better and cheaper than anything they can purchase. Brett ___ --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] Native MOH - Convert mp3 to ulaw
On 3/22/2006, "John Novack" <[EMAIL PROTECTED]> wrote: > > Douglas Garstang wrote: > >> Good grief. Considering all the libraries and requirements, it'd almost >> be easier to find some windows software to do this. >> > Wavepad works well, without complaining about libraries, and you can > even edit. listen to the results and back out,if need be. > Harder to use for those who aren't sighted, though Of course he can just use mpg123 and convert it to a wave. Then use sox to convert it to whatever... Brett ___ --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] How to disable event_log?
On 3/27/2006, "Roger Schreiter" <[EMAIL PROTECTED]> wrote: >Hi, > >how can I disable event_log in order to reduce >hard disk activity? > >I can't find any hints in conf files. Roger. check logger.conf under the general heading. ; This determines whether or not we log generic events to a file ; (defaults to yes). ;event_log = no Brett ___ --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] Looking for a good VoIP Provider in the UK-
On 4/17/2006, "Maxx Lobo" <[EMAIL PROTECTED]> wrote: > > Any recommendations for a VoIP provider in the UK? > > I have a few guys in a field office in the UK with SIP phones and a VPN > tunnel back to a working Asterisk setup in the US. The Asterisk setup > has an IAX trunk with TelaSIP/VoipXpress with local DiD's for US > offices, so they can call vendors, customers etc in the US at local > rates. I'd like to get the same thing for the UK, so that UK customers > can call them as a local call AND they can dial out UK numbers as local > calls. The obvious side benefit would be that US employees could call UK > customers and vendors as a local call as well. > > I've looked at Telappliant, VoipTalk and PipeCall so far, and I'd like > to get some feedback before going with one or the other. > > I'd be grateful for any opinions on the quality of (these, other) > services, how responsive they are to problems, and if they are as easy > to setup with Asterisk as TelaSIP. Recommendations are appreciated, of > course. > > Thanks- > >--Maxx Hey Maxx! Get on over to voipuser.org and read the forums. They are based out of London I believe and if you need any info on VoIP in the UK they are an extremely good source. You probably don't want to use their 'service' but they are extremely knowledgable. Brett ___ --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] chat with asterisk
On 6/19/2006, "issam" <[EMAIL PROTECTED]> wrote: >Hello >How can I do a vocal chat with asterisk >Issam >Thanks Issam - check out MeetMe and app_conference - probably well documented on the wiki. Brett ___ --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] linux journal article on asterisk
arrggh, sorry about the website. It should be back up now... Yes, I hang out here, but don't talk much, since I am really busy right now... I am *really* close to you, I work in Redmond and live in Bellevue :) --- calvis <[EMAIL PROTECTED]> wrote: > Thanks for the link. > > This is an interesting article on Asterisk. I was > hoping to send him some > kudos, but his website isn't working at > http://www.bschwarz.com/. And I > just noticed the guy lives near me! > > Does anyone know if he hangs out on the list? > > Charles > Internet Technology Group, Inc. > Redmond, WA > > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On > Behalf Of Tony Kava > Sent: Wednesday, January 14, 2004 7:56 AM > To: '[EMAIL PROTECTED]' > Subject: RE: [Asterisk-Users] linux journal article > on asterisk > > > > > For anybody who didn't know there is an > article on asterisk in > > > > February's Linux Journal. > > > > > > Can you please provide a link to this article? > > > Franz> From: [EMAIL PROTECTED] > > Here's the link (I believe): > > http://www.linuxjournal.com/article.php?sid=6769 > > -- > Tony Kava > Senior Network Administrator > Pottawattamie County, Iowa __ Do you Yahoo!? Yahoo! Hotjobs: Enter the "Signing Bonus" Sweepstakes http://hotjobs.sweepstakes.yahoo.com/signingbonus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] X100P and NTL (ex Cable + Wireless)
Firstly, let me just say I am new to asterisk and if anything I've said is covered in an FAQ or in previous posts I apologise but I have tried searching and I've attempted a few of the things I found but they didn't help. Has anybody got any experience using an X100P on an NTL phone line in the UK (I'm in an ex Cable & Wireless area if that makes any difference). The problem I'm having (and judging by the number of references to it I've found searching it is a common one) is getting * to detect when the line has been hung up. It doesn't matter if it comes through to a person directly as they can just hang that phone up, but when it hits voicemail, and it sits there for two minutes recording an empty message, and then emails it to the person it can be a bit annoying! There are a couple of possible things that I realize could be causing it, the most major being that the phone wiring to the port that asterisk is plugged into is a bit dodgy and I think there is a fair amount of interference which I suspect could be annoying it a bit. The other possible problem is that when someone hangs up, the tone that comes down the phone line to indicate that it has been hung up lasts for about 4 seconds, and then the line just goes silent - would this mean that the busy detect mechanism would not detect that it has been hung up? Thanks in advance, Alex Brett [EMAIL PROTECTED] +44 (0)870 744 2170 http://www.loho.co.uk/ ___ 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] X100P and NTL (ex Cable + Wireless)
Stephen Davies wrote: >Hi Alex, > >Indeed the call end termination doesn't work on an NTL line. I'm not >so sure it works too well on other lines either. > >I did some work a while back to add detection of the UK busy/hangup >signal on the line, but I never got it working well enough to depend >on it. The problem is that it is a single frequency tone. (The US >one is dual-tone). Women's voices used to sometimes trigger my >detector - causing hangups. > >The main practical issue is with voicemail, as you say. > >My final solution was to switch to ISDN. > >Steve I would be interested in seeing this code, as I have found that the hangup signal I get lasts for an exact amount of time before the line goes silent, so if I time it precisely, I may be able to adapt the code to only trigger if the tone lasts greater than a certain amount but less than another amount, which should prevent it triggering on voices (as it is about 4 seconds and the likelihood of that tone coming from a female voice and lasting 4 seconds is slim). Thanks, Alex Brett [EMAIL PROTECTED] +44 (0)870 744 2170 http://www.loho.co.uk/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with x-ten lite
The problem could be to do with the silence surpression feature in X-ten Lite. If you go into Advanced System Settings, Audio Settings, Silence Settings, you should have Transmit Silence set to Yes as Asterisk has a compatibility issue with the way it tries to surpress transmitting the silence (or at least it did when I first used it and I'm assuming it still does, if I'm wrong please feel free to correct me...). Shad Mortazavi wrote: I prefer the look and feel of the x-ten lite. However, when ever I use my x-ten lite I get a lot of breakup in my communication. E.g. I will play some hold music, and every 5-6 seconds I drop some packets. I don’t have the same issue with SJPhone. I’m sure this is a configuration issues, but I can work out where? Can someone point me in the right directions? Thanks and Regards Shad Mortazavi --- **Nexus Technical Manager** n|m Nexus Management Inc Neutral Bay Sydney Hope this helps, Alex Brett [EMAIL PROTECTED] +44 (0)870 744 2170 http://www.loho.co.uk/ ___ 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] Modems compatible with NTL caller id
I'm looking at using a modem to provide caller ID info on my NTL line following the steps in the article posted by Darren Poulson: http://www.22balmoralroad.net/modules.php?name=Sections&op=viewarticle&artid=1 I was wondering if anybody had experience of using any modems (can be pci/isa/external) with an ntl line (in an ex. Cable & Wireless area if it makes a difference) and would know which modems will pick up the caller id. Failing anybody having direct experience, does anybody know what caller id standard NTL use so I can find a modem that supports that standard or even maybe get Asterisk working with it directly (I haven't actually looked yet to see if Caller ID is enabled on the line I'm using so it could be as simple as just getting ntl to switch it on). Thanks, Alex Brett [EMAIL PROTECTED] +44 (0)870 744 2170 http://www.loho.co.uk/ ___ 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] PSTN Call drops randomly
Do you need the busy detect feature as this often can cause random drop outs. Try changing busydetect=yes to busydetect=no in zapata.conf. Assuming that asterisk still works (i.e. it detects when people hang up and when lines are busy (most lines will do it ok but some won't (e.g. ntl cable in the UK from personal experience)) then that *should* solve your problem, if not then something more serious is going on. Shahid Mahmood wrote: Dear List members, After succesfully installing the * on a couple of systems, and putting them on test, I observed that there is an intermittent call drop on PSTN line. The systems are - Dell Optiplex P3/500MHz/128MB - Built-in ethernet - 1 X100P (Motorolla chip) card on PCI - 10G HDD etc. - Asterisk April 17 CVS. - 2 Mediatrix FXS ATA (4 phones) - 2 Grandstream phones. - sip.conf, zaptel.comnf and zapata.conf included below Also let me know what do I need to "turn on" to get fine details about the event when it happens. Any help will be greatly appreciated. Regards. -shahid < SNIP > -- Alex Brett [EMAIL PROTECTED] +44 (0)870 744 2170 http://www.loho.co.uk/ ___ 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] Using Exchange to send voicemail message
As far as I know asterisk won't talk directly to an SMTP server. All you have to do is set sendmail to relay all mail through your exchange server (the DS directive in sendmail.cf) and then you are sorted, the only difference between it directly talking to the exchange server is one extra 'Received from' header on the e-mail. Paul Tyreman wrote: An of course, its SMTP that I need. I don't want sendmail to send mail to the exchange server, I want to use the exchange server to send the mail in the first place ! What I want to do is forget about sendmail, and make an account on the exchange server for asterisk to send mail from. Can that be done ? Thanks, Paul. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert Hajime Lanning Posted At: 25 April 2004 19:40 Posted To: Asterisk-Users Conversation: [Asterisk-Users] Using Exchange to send voicemail message Subject: Re: [Asterisk-Users] Using Exchange to send voicemail message IMAP and POP3 are used for the MUA to get access to a mailbox. They are not used for delivering messages to a mailbox, but for reading message out of a mailbox. What you are looking for, is an SMTP gateway. Sendmail is an SMTP MTA that can be configured to send the email (via SMTP/ESMTP) to the Exchange server. All you really need to do is have the DNS MX records for foo.com pointing to your Exchange server. Then, in voicemail.conf you would have the email address set to [EMAIL PROTECTED] Of course, change foo.com to whatever your domain, for the Exchange server, is. And, make sure you have the SMTP connector configured for Exchange. > Hi, > > I run a local exchange server and would like asterisk to send > voicemail notification messages via exchange. > > I have had a look at the voicemail.conf file, but I can't see how I > would go about configuring it to use an account set up on exchange ? > The exchange account would have both POP3 and IMAP access, so how can > I tell Asterisk to use the exchange account rather then sendmail ? > > Thanks, Paul. -- END OF LINE -MCP ___ 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 -- Alex Brett [EMAIL PROTECTED] +44 (0)870 744 2170 http://www.loho.co.uk/ ___ 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] dial application - continue in context
Title: Message Hi All, I'm tring to do some DB operations before and after a call. I see the 'g' option in dial to continue in context if the destination hangs up, but what if the originator hangs up? Basically I do a DB get/put before the call is placed. After the call is completed I want to do another get/put; however the dial application dies when the originator hangs up. Any way to get around this? Perhaps there is a better way to do this (AGI?) Thanks, Brett
RE: [Asterisk-Users] dial application - continue in context
Hi Phillip, It needs to occur right after the call. I'm tring to apply a sort of fromdomain call limit. So I need to keep track of how many are currently active -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Philipp von Klitzing Sent: Saturday, May 22, 2004 6:29 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] dial application - continue in context Hi! > I'm tring to do some DB operations before and after a call. I see the > 'g' > option in dial to continue in context if the destination hangs up, but > what if the originator hangs up? You either need to run a CRON job for this clean up, or do that at the beginning of the next call - whatever suits you better. Note: The h extension is not reliable enough to solve your problem. Cheers, Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users 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] Sip Registration Problem
Title: Message Hi All, I had an unusual problem today; I'm sure it's a configuration problem. I had 2 phones behind a nat device and I had qualify=300 in both extensions config. The device I was talking to was an edgewater traffic shaper,/Sip Proxy. Since it is acting as a sip proxy, it was ignoring the OPTIONS messages that * was sending, and thus * interpreted that as the extensions being down. I removed the qualify lines and sip reload [ed]. The extension still showed up as "UNREACHABLE" instead of "UNMONITORED". I had to do a full restart to get it to stop sending the OPTIONS messages. What did I do wrong here? How can I make a change to qualify without restarting? Thanks all, Brett
RE: [Asterisk-Users] Sip Registration Problem
How will this effect a live system? No new calls? Or will it terminate exisiting calls? I'll have a chat with the vendor regarding the OPTIONS reply.. It certainly does sesem like it should reply with something.. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Olle E. Johansson Sent: Tuesday, May 25, 2004 1:13 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Sip Registration Problem Karl Brose wrote: > Btw, Ignoring OPTIONS is not a valid option (:-) whether sip proxy or > not, Asterisk doesn't do it correctly either. > The host should respond with 200/OK if the call >could< succeed > theoretically if it were an INVITE or else it should send a > 404 or maybe a 487(? hmm, have to look) see the RFC for details. Interesting, didn't know that. Where in the RFC? >> I removed the qualify lines and sip reload [ed]. The extension still >> showed up as "UNREACHABLE" instead of "UNMONITORED". I had to do a >> full restart to get it to stop sending the OPTIONS messages. >> >> What did I do wrong here? How can I make a change to qualify without >> restarting? If a peer is registred at reload/sip reload, it will not change. You have to unload the sip module and reload it or restart asterisk to change the configuration of a registred, i.e. active, peer. /O ___ 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] Billing, Radius, anyone?
Title: Message Hi All, I found app_radius and cdr_radius, but I'm not sure how to use them. They are poorly documented. There are a few open source billing systems (trabas) out there that use radius as the cdr method and I'm wondering how this can be integrated with *. Just adding cdr_radius, like cdr_mysql doesn't work.. Does anyone have any ideas? Anyone know of a billing system that can just use the mysql cdr format? or perhaps import scripts to use it?? The wiki didn't seem to have a lot of assistance here.. Thanks! -Brett
[Asterisk-Users] TE110P yellow errors
Hi All, I have a TE110P in E1 mode, in a dell poweredge 250. The 30 channel E1 supplied is from a telco in Australia, with the following in my zaptel.conf: span=1,1,0,ccs,hdb3 bchan=1-15 dchan=16 bchan=17-31 (yes the provider has NOcrc4 for some reason) After installing the latest cvs libpri and zaptel, I successfully loaded the kernel modules. The card is verified as not sharing an interrupt with anything else. The problem is I continuously get yellow errors, the IRQ missed counter goes up, and the light on the card blinks b/w red and green. I have verified the cabling, as it works fine in the E1 port of a cisco router below it. Any ideas most welcome thanks. TIA ___ 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] TE110P yellow errors
Hi All, I fixed the missed IRQ's with the following reference: http://lists.digium.com/pipermail/asterisk-users/2004-July/054712.html At 01:09 AM 26/01/2005, you wrote: Hi All, I have a TE110P in E1 mode, in a dell poweredge 250. The 30 channel E1 supplied is from a telco in Australia, with the following in my zaptel.conf: span=1,1,0,ccs,hdb3 bchan=1-15 dchan=16 bchan=17-31 (yes the provider has NOcrc4 for some reason) After installing the latest cvs libpri and zaptel, I successfully loaded the kernel modules. The card is verified as not sharing an interrupt with anything else. The problem is I continuously get yellow errors, the IRQ missed counter goes up, and the light on the card blinks b/w red and green. I have verified the cabling, as it works fine in the E1 port of a cisco router below it. Any ideas most welcome thanks. TIA ___ 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 All the best, Brett Murphy Director, Alphalink (Australia) PTY LTD ph: +61 3 9495-9000 fax: +61 3 9486-6822 email: [EMAIL PROTECTED] The contents of this message may not be quoted, copied, reproduced or published in part or in whole, without the written authorization of Brett Murphy, Director, Alphalink (Australia) Pty Ltd. ___ 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] Getting a Wildcard TE110P working on E1's in Australia
Hello All, I have got my TE110P working at the hardware level, turned out to be a dodgy cable causing the Yellow errors in zttool. However, now I am getting yellow errors in asterisk, but zttool shows nothing out of the ordinary. here is my current config, and some asterisk console errors, any suggestions most welcome. zaptel.conf span=1,1,0,ccs,hdb3 bchan=1-15 dchan=16 bchan=17-31 zapata.conf [channels] signalling=pri_cpe switchtype=euroisdn language=en context=sip channel => 1-15 channel => 17-31 echocancel= yes echocancelwhenbridged = yes echotraining = yes group = 1 usecallingpres= yes console errors: Jan 26 19:39:45 NOTICE[2158]: chan_zap.c:7381 pri_dchannel: PRI got event: Alarm (4) on Primary D-channel of span 1 Jan 26 19:39:45 WARNING[2158]: chan_zap.c:1925 pri_find_dchan: No D-channels available! Using Primary on channel anyway 16! Jan 26 19:39:45 WARNING[2159]: chan_zap.c:5642 handle_init_event: Detected alarm on channel 1: Yellow Alarm Jan 26 19:39:45 WARNING[2159]: chan_zap.c:5642 handle_init_event: Detected alarm on channel 2: Yellow Alarm Jan 26 19:39:45 WARNING[2159]: chan_zap.c:5642 handle_init_event: Detected alarm on channel 3: Yellow Alarm Jan 26 19:39:45 WARNING[2159]: chan_zap.c:5642 handle_init_event: Detected alarm on channel 4: Yellow Alarm Jan 26 19:39:45 WARNING[2159]: chan_zap.c:5642 handle_init_event: Detected alarm on channel 5: Yellow Alarm Jan 26 19:39:45 WARNING[2159]: chan_zap.c:5642 handle_init_event: Detected alarm on channel 6: Yellow Alarm Jan 26 19:39:45 WARNING[2159]: chan_zap.c:5642 handle_init_event: Detected alarm on channel 7: Yellow Alarm Jan 26 19:39:45 WARNING[2159]: chan_zap.c:5642 handle_init_event: Detected alarm on channel 8: Yellow Alarm Jan 26 19:39:45 WARNING[2159]: chan_zap.c:5642 handle_init_event: Detected alarm on channel 9: Yellow Alarm Jan 26 19:39:45 WARNING[2159]: chan_zap.c:5642 handle_init_event: Detected alarm on channel 10: Yellow Alarm Jan 26 19:39:45 WARNING[2159]: chan_zap.c:5642 handle_init_event: Detected alarm on channel 11: Yellow Alarm Jan 26 19:39:45 WARNING[2159]: chan_zap.c:5642 handle_init_event: Detected alarm on channel 12: Yellow Alarm Jan 26 19:39:45 WARNING[2159]: chan_zap.c:5642 handle_init_event: Detected alarm on channel 13: Yellow Alarm Jan 26 19:39:45 WARNING[2159]: chan_zap.c:5642 handle_init_event: Detected alarm on channel 14: Yellow Alarm Jan 26 19:39:45 WARNING[2159]: chan_zap.c:5642 handle_init_event: Detected alarm on channel 15: Yellow Alarm Jan 26 19:39:45 WARNING[2159]: chan_zap.c:5642 handle_init_event: Detected alarm on channel 17: Yellow Alarm Jan 26 19:39:45 WARNING[2159]: chan_zap.c:5642 handle_init_event: Detected alarm on channel 18: Yellow Alarm Jan 26 19:39:45 WARNING[2159]: chan_zap.c:5642 handle_init_event: Detected alarm on channel 19: Yellow Alarm Jan 26 19:39:45 WARNING[2159]: chan_zap.c:5642 handle_init_event: Detected alarm on channel 20: Yellow Alarm Jan 26 19:39:45 WARNING[2159]: chan_zap.c:5642 handle_init_event: Detected alarm on channel 21: Yellow Alarm Jan 26 19:39:45 WARNING[2159]: chan_zap.c:5642 handle_init_event: Detected alarm on channel 22: Yellow Alarm Jan 26 19:39:45 WARNING[2159]: chan_zap.c:5642 handle_init_event: Detected alarm on channel 23: Yellow Alarm Jan 26 19:39:45 WARNING[2159]: chan_zap.c:5642 handle_init_event: Detected alarm on channel 24: Yellow Alarm Jan 26 19:39:45 WARNING[2159]: chan_zap.c:5642 handle_init_event: Detected alarm on channel 25: Yellow Alarm Jan 26 19:39:45 WARNING[2159]: chan_zap.c:5642 handle_init_event: Detected alarm on channel 26: Yellow Alarm Jan 26 19:39:45 WARNING[2159]: chan_zap.c:5642 handle_init_event: Detected alarm on channel 27: Yellow Alarm Jan 26 19:39:45 WARNING[2159]: chan_zap.c:5642 handle_init_event: Detected alarm on channel 28: Yellow Alarm Jan 26 19:39:45 WARNING[2159]: chan_zap.c:5642 handle_init_event: Detected alarm on channel 29: Yellow Alarm Jan 26 19:39:45 WARNING[2159]: chan_zap.c:5642 handle_init_event: Detected alarm on channel 30: Yellow Alarm Jan 26 19:39:45 WARNING[2159]: chan_zap.c:5642 handle_init_event: Detected alarm on channel 31: Yellow Alarm Jan 26 19:39:45 NOTICE[2158]: chan_zap.c:7381 pri_dchannel: PRI got event: No more alarm (5) on Primary D-channel of span 1 Jan 26 19:39:45 WARNING[2158]: chan_zap.c:1925 pri_find_dchan: No D-channels available! Using Primary on channel anyway 16! Jan 26 19:39:45 NOTICE[2159]: chan_zap.c:5637 handle_init_event: Alarm cleared on channel 1 Jan 26 19:39:45 NOTICE[2159]: chan_zap.c:5637 handle_init_event: Alarm cleared on channel 2 Jan 26 19:39:45 NOTICE[2159]: chan_zap.c:5637 handle_init_event: Alarm cleared on channel 3 Jan 26 19:39:45 NOTICE[2159]: chan_zap.c:5637 handle_init_event: Alarm cleared on channel 4 Jan 26 19:39:45 NOTICE[2159]: chan_zap.c:5637 handle_init_event: Alarm cleared on channel 5 Jan 26 19:39:45 NOTICE[2159]: chan_zap.c:5637 handle_init_event: Alarm cleared on c
[Asterisk-Users] Reccomendation for reliable handsets
Hi there I'm sure this question has been raised a number of times before, but unfortunately I do not have direct access to the archives I am about to roll out Asterisk to a few companies and would like to hear your experiences about the various handsets/phones that are Asterisk compatible I am primarily looking for 2 options, the first being a cheaper model which will provide reliability whilst still maintaining a reasonable feature set, and a reliable model from the more expensive range with more features But the definite focus here is on reliability and ease of maintenance Any help or advice would be greatly appreciated; I would really like to hear your experiences/recommendations Cheers Gary ___ 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] Reccomendation for reliable handsets
Sorry to move this up the list again, but does anybody have any advice on this -Original Message- From: Brett, Gary [mailto:[EMAIL PROTECTED] Sent: 02 February 2005 10:49 To: 'asterisk-users@lists.digium.com' Subject: [Asterisk-Users] Reccomendation for reliable handsets Hi there I'm sure this question has been raised a number of times before, but unfortunately I do not have direct access to the archives I am about to roll out Asterisk to a few companies and would like to hear your experiences about the various handsets/phones that are Asterisk compatible I am primarily looking for 2 options, the first being a cheaper model which will provide reliability whilst still maintaining a reasonable feature set, and a reliable model from the more expensive range with more features But the definite focus here is on reliability and ease of maintenance Any help or advice would be greatly appreciated; I would really like to hear your experiences/recommendations Cheers Gary ___ 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 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] Web based Asterisk management tool
Hi there I am new to Asterisk and am looking for a web based management tool, for managers to manage hunt groups, extensions etc and for user to have access to there own phone features. I have seen there are a number of commercial tools available for this, but I presume there are some freeware options too I noticed one that I like at http://www.thirdlane.com/screenshots.htm but I am assuming this is just a freeware product that has been re-badged so to speak. If any body can give me some suggestions that would be great Regards Gary ___ 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] Web based Asterisk management tool
Thanks for the responses, isn't [EMAIL PROTECTED] an pre-configured distribution for small systems?? I am looking for an open source web management tool to use on any size asterisk server (even ones that are already up and running) the user base could be anything between small and large with many external lines, Ive looked at AMP, is it free ? and are there any alternatives or is AMP the only open source web management tool ? -Original Message- From: dean collins [mailto:[EMAIL PROTECTED] Sent: 09 February 2005 15:05 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Web based Asterisk management tool That would be the AMP database, I don't know. Ping the amp list and find out. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Daniel Eboa Sent: Wednesday, February 09, 2005 9:47 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Web based Asterisk management tool How big can be [EMAIL PROTECTED] user data base? Can it handle 1000s of users ? Regards. Daniel. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of dean collins Sent: mercredi 9 février 2005 15:42 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Web based Asterisk management tool Hi Gary, do a search for [EMAIL PROTECTED] the iso is available for download at sourceforge and does exactly what you are looking for. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brett, Gary Sent: Wednesday, February 09, 2005 8:01 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Web based Asterisk management tool Hi there I am new to Asterisk and am looking for a web based management tool, for managers to manage hunt groups, extensions etc and for user to have access to there own phone features. I have seen there are a number of commercial tools available for this, but I presume there are some freeware options too I noticed one that I like at http://www.thirdlane.com/screenshots.htm but I am assuming this is just a freeware product that has been re-badged so to speak. If any body can give me some suggestions that would be great Regards Gary ___ 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 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 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 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 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] Web based Asterisk management tool
Thanks Dean, you say that " [EMAIL PROTECTED] is just an automated way of installing AMP and FOP and Web Meetme." But from the installation instructions I have read, you download an ISO image that installs a linux distro for you (destroying current install) and then configures itself for use I do not want to overwrite any existing systems, here is a quote from the installation page of [EMAIL PROTECTED] 1) Burn [EMAIL PROTECTED] iso to a blank CD 2) Boot your Asterisk PC with the CD and press enter NOTE: This will erase all data on the hard drive of the PC!!! Etc etc All I want is the web config tool ! Apologies if I am misunderstanding you here, as I say I am quite new to this and need to get up to speed fast For an Admin only web based product, is AMP my only option ?? Cheers again -Original Message- From: dean collins [mailto:[EMAIL PROTECTED] Sent: 09 February 2005 16:48 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Web based Asterisk management tool You need to go back and reread. It is just pretty much an asterisk configuration tool (ok some minor things in the backend but it's the best out there). AMP is available for free download but they make their money by offering support. [EMAIL PROTECTED] is just an automated way of installing AMP and FOP and Web Meetme. If you really have a need to support thousands of extensions as you suggest then you should really go back and learn how to program asterisk with a database yourself from scratch. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brett, Gary Sent: Wednesday, February 09, 2005 11:17 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Web based Asterisk management tool Thanks for the responses, isn't [EMAIL PROTECTED] an pre-configured distribution for small systems?? I am looking for an open source web management tool to use on any size asterisk server (even ones that are already up and running) the user base could be anything between small and large with many external lines, Ive looked at AMP, is it free ? and are there any alternatives or is AMP the only open source web management tool ? -Original Message- From: dean collins [mailto:[EMAIL PROTECTED] Sent: 09 February 2005 15:05 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Web based Asterisk management tool That would be the AMP database, I don't know. Ping the amp list and find out. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Daniel Eboa Sent: Wednesday, February 09, 2005 9:47 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Web based Asterisk management tool How big can be [EMAIL PROTECTED] user data base? Can it handle 1000s of users ? Regards. Daniel. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of dean collins Sent: mercredi 9 février 2005 15:42 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Web based Asterisk management tool Hi Gary, do a search for [EMAIL PROTECTED] the iso is available for download at sourceforge and does exactly what you are looking for. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brett, Gary Sent: Wednesday, February 09, 2005 8:01 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Web based Asterisk management tool Hi there I am new to Asterisk and am looking for a web based management tool, for managers to manage hunt groups, extensions etc and for user to have access to there own phone features. I have seen there are a number of commercial tools available for this, but I presume there are some freeware options too I noticed one that I like at http://www.thirdlane.com/screenshots.htm but I am assuming this is just a freeware product that has been re-badged so to speak. If any body can give me some suggestions that would be great Regards Gary ___ 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 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 mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.
RE: [Asterisk-Users] Reccomendation for reliable handsets
Thanks Mark I am definitely interested in the budgetone 102 but am a little concerned about the 10mbit only Ethernet ports !! From what I have read, these are relatively new models and I like the addition of a second port to daisy chain your PC from the same network connection, however why 10mbits and not 100mbits ??, I would have thought this would be a minimum these days, I don't know anyone who still runs 10mbits to the desktop, and im not too happy about bottlenecking my customers fast Ethernet network with these phones A real shame really. Does anybody know if Grandstream will be addressing this or indeed if they have any current models with at least 100mbit ports Regards -Original Message- From: Mark Benson [mailto:[EMAIL PROTECTED] Sent: 03 February 2005 14:27 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Reccomendation for reliable handsets I have been using an IN1002 generic handset (supposed to be an unbranded cisco copy but I am skeptical) for a few months (6months+) now, and it seems pretty stable - however I haven't found a reliable supplier Also there is almost no support for them.. I have switched to the grandstream budgetone 102 and they seem pretty good too. You can pretty much plug in and forget it with both phones. They do lock up occasionally (once a month to once every 3 months). I have yet to upgrade the firmware on the grandstreams... Mark Brett, Gary wrote: >Sorry to move this up the list again, but does anybody have any advice on >this > >-Original Message- >From: Brett, Gary [mailto:[EMAIL PROTECTED] >Sent: 02 February 2005 10:49 >To: 'asterisk-users@lists.digium.com' >Subject: [Asterisk-Users] Reccomendation for reliable handsets > >Hi there > >I'm sure this question has been raised a number of times before, but >unfortunately I do not have direct access to the archives > >I am about to roll out Asterisk to a few companies and would like to hear >your experiences about the various handsets/phones that are Asterisk >compatible > >I am primarily looking for 2 options, the first being a cheaper model which >will provide reliability whilst still maintaining a reasonable feature set, >and a reliable model from the more expensive range with more features > >But the definite focus here is on reliability and ease of maintenance > > > >Any help or advice would be greatly appreciated; I would really like to hear >your experiences/recommendations > >Cheers >Gary > > > > > > > >___ >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 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 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 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] Digium Cards connecting to BT
Hi there Just a general question, has anybody experienced any problems with any Digium telephony cards in the UK, specifically with BT (British Telecom) lines. I just want to make sure there are no compatibility issues before purchasing cards, (mainly TDM400P's) Any comments would be greatly appreciated Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Reccomendation for reliable handsets
Bob, Thanks for your reply, im not sure what top posting is, but I have been on holiday and am simply replying to a response that was given to my original question, If you could explain to me how I go about continuing the thread it would be much appreciated, with regards to your reply, I am indeed daisy chaining to the PC, hence my post point regarding bottlenecking the 100mbits to the desktop I just find it hard to understand the point of releasing a phone with a 2 port hub yet still limiting to ports to 10mbits, anyway , I take it there are no alternatives from budgetone, so I will have to at other low cost models Thanks all -Original Message- From: Bob Goddard [mailto:[EMAIL PROTECTED] Sent: 14 February 2005 14:11 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Reccomendation for reliable handsets On Monday 14 February 2005 13:00, Brett, Gary wrote: > Thanks Mark > > I am definitely interested in the budgetone 102 but am a little concerned > about the 10mbit only Ethernet ports !! From what I have read, these are > relatively new models and I like the addition of a second port to daisy > chain your PC from the same network connection, however why 10mbits and not > 100mbits ??, I would have thought this would be a minimum these days, I > don't know anyone who still runs 10mbits to the desktop, and im not too > happy about bottlenecking my customers fast Ethernet network with these > phones > > A real shame really. Does anybody know if Grandstream will be addressing > this or indeed if they have any current models with at least 100mbit ports Please do not top post. I don't think there is a single IP phone which can flood a 10Mbps port. You do not need 100Mbps on a phone unless it has a passthrough to a PC. Let's see, using a 64Kbps codec and being generous, will use 100Kbps on the wire. Assuming the 10Mbps port can reliably run at 8Mbps, that means the phone would have to have 8 * 1000 / 100 = 80 concurrent RTP streams. Can anyone see anything wrong with my rough calculations? B > -Original Message- > From: Mark Benson [mailto:[EMAIL PROTECTED] > Sent: 03 February 2005 14:27 > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Reccomendation for reliable handsets > > I have been using an IN1002 generic handset (supposed to be an unbranded > cisco copy but I am skeptical) for a few months (6months+) now, and it > seems pretty stable - however I haven't found a reliable supplier Also > there is almost no support for them.. > > I have switched to the grandstream budgetone 102 and they seem pretty > good too. You can pretty much plug in and forget it with both phones. > They do lock up occasionally (once a month to once every 3 months). I > have yet to upgrade the firmware on the grandstreams... > > Mark > > Brett, Gary wrote: > >Sorry to move this up the list again, but does anybody have any advice on > >this > > > >-Original Message- > >From: Brett, Gary [mailto:[EMAIL PROTECTED] > >Sent: 02 February 2005 10:49 > >To: 'asterisk-users@lists.digium.com' > >Subject: [Asterisk-Users] Reccomendation for reliable handsets > > > >Hi there > > > >I'm sure this question has been raised a number of times before, but > >unfortunately I do not have direct access to the archives > > > >I am about to roll out Asterisk to a few companies and would like to hear > >your experiences about the various handsets/phones that are Asterisk > >compatible > > > >I am primarily looking for 2 options, the first being a cheaper model > > which will provide reliability whilst still maintaining a reasonable > > feature set, and a reliable model from the more expensive range with more > > features > > > >But the definite focus here is on reliability and ease of maintenance > > > > > > > >Any help or advice would be greatly appreciated; I would really like to > > hear > > >your experiences/recommendations > > > >Cheers > >Gary > > > > > > > > > > > > > > > >___ > >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 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/ma
[Asterisk-Users] asterisk@home in production env
Hi there I just wanted to know what the difference between [EMAIL PROTECTED] and manually built boxes actually is ?? What makes [EMAIL PROTECTED] a "home" system ? Is it not a good idea to run [EMAIL PROTECTED] then modify/tweak it to use in a production environment ??, if so why not, would somebody be able to explain this to me, it seems to have a "hobbiest" tag associated with it, and i just wanted to know the difference cheers Gary ___ 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] Voicemail / Dial command issue
Hi,
I have a load of IAX extensions, which I'm trying to set up a standard
macro to dial them, which gives unavailable or busy voicemail if there
is no answer or the phone is in use respectively.
The macro I have at the moment is:
; std-exten macro, ${ARG1} = Device to call, ${ARG2} = voicemail box
[macro-std-exten]
; Call the user for 20 seconds
exten => s,1,Dial(${ARG1},20,tr)
exten => s,2,Goto(s-${DIALSTATUS},1)
; If unavailable, go to voicemail
exten => s-NOANSWER,1,Voicemail(u${ARG2})
exten => s-NOANSWER,2,Hangup
; If busy, go to voicemail
exten => s-BUSY,1,Voicemail(b${ARG2})
exten => s-BUSY,2,Hangup
And the bit of the dial plan that calls this is:
exten => 2002,1,Macro(std-exten,IAX2/2002,2001)
(Ignore the fact I'm using voicemail box 2001, its just as I haven't set
2002 up yet).
Now, from what I can tell this should be fine. The noanswer bit works
perfectly, if I call it and don't answer, I hit voicemail with the
unavailable message, however, if the phone is in use and it gets called,
it also hits the unavailable message rather than the busy message. I've
done a bit of digging and as far as I can tell the Dial command seems to
be ignoring the response from the IAX, take this for example (the
receiving phone (2002) has dialled an extension I've set up that just
plays hold music, and is at x.x.x.x (IPs masked for security reasons).
The dialling phone is at y.y.y.y):
This is extension 2002 calling hold music, nothing strange here:
-- Accepting AUTHENTICATED call from x.x.x.x:
> requested format = ilbc,
> requested prefs = (),
> actual format = ilbc,
> host prefs = (),
> priority = mine
-- Executing MusicOnHold("IAX2/[EMAIL PROTECTED]", "") in new stack
-- Started music on hold, class 'default', on IAX2/[EMAIL PROTECTED]
This is the call coming in from the second phone to call 2002:
-- Accepting AUTHENTICATED call from y.y.y.y:
> requested format = ilbc,
> requested prefs = (),
> actual format = ilbc,
> host prefs = (ilbc),
> priority = mine
-- Executing Macro("IAX2/[EMAIL PROTECTED]",
"std-exten|IAX2/2002|2001") in new stack
-- Executing Dial("IAX2/[EMAIL PROTECTED]", "IAX2/2002|20|tr") in
new stack
-- Called 2002
This seems to make sense, here is the IAX handler saying that the call
was rejected by x.x.x.x as it is In call:
Mar 27 20:26:56 WARNING[25053]: chan_iax2.c:6735 socket_read: Call
rejected by x.x.x.x: In call
-- Hungup 'IAX2/2002-6'
This is what confuses me, as it seems the Dial application (or is it
something else?) is saying that it is a no answer rather than a busy!
== No one is available to answer at this time (1:0/0/0)
At this point it has obviously set the ${DIALSTATUS} to NOANSWER and
hence the rest of the call follows this:
-- Executing Goto("IAX2/[EMAIL PROTECTED]", "s-NOANSWER|1") in
new stack
-- Goto (macro-std-exten,s-NOANSWER,1)
-- Executing VoiceMail("IAX2/[EMAIL PROTECTED]", "su2001") in new
stack
-- Playing '/var/spool/asterisk/voicemail/default/2001/unavail'
(language 'en')
== Spawn extension (macro-std-exten, s-NOANSWER, 1) exited non-zero
on 'IAX2/[EMAIL PROTECTED]' in macro 'std-exten'
== Spawn extension (internal, 2002, 1) exited non-zero on
'IAX2/[EMAIL PROTECTED]'
-- Hungup 'IAX2/[EMAIL PROTECTED]'
And finally I hang up extensions 2002:
-- Stopped music on hold on IAX2/[EMAIL PROTECTED]
== Spawn extension (internal, 500, 1) exited non-zero on
'IAX2/[EMAIL PROTECTED]'
-- Hungup 'IAX2/[EMAIL PROTECTED]'
In case it is needed, relevant portions of iax.conf:
[general]
bandwidth=high
allow=all
disallow=lpc10
jitterbuffer=no
notransfer=yes
[2002]
type=friend
username=2002
secret=**
host=dynamic
notransfer=yes
context=internal
If anybody could shed any light on this strange behaviour, it would be
much appreciated, as otherwise I'm just going to have to roll with one
voicemail message which isn't a huge problem, but slightly annoying.
Thanks in advance,
Alex Brett
[EMAIL PROTECTED]
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[Asterisk-Users] ISDN question
Hi there I wonder if anybody has had this problem before, I am running Asterisk v1.06 stable on FC1 with a digium TE110p card (I am in the UK so am using E1 through British Telecom with my * server as the PRI_CPE). The strange thing is, everything is fine, no problems with asterisk, the card comes up fine I can receive calls, but the problem comes when making calls, when I dial out, it gets to the British Telecom exchange and brings back the BT message "The number you have dialled has not been recoginised" . so the first thing I did was to make sure that the number was the full telephone number, and indeed it is, my dial plan does not prepend any digits to outbound calls so there is no reason to doubt that I am sending the full string to BT, even my CLI has the full number ie Zap/g1/02087775566 , very strange. I am fortunate enough to have at my disposal a spare E1 card on a 3com NBX Voip Switch, so as a test ran a back to back E1 crossover between the 3Com and the * Server. I configured the dial plan on the 3com to forward whatever it received directly out to the telco (with no digit manipulation, effectivly bridging the call) and hey presto, the destination phone rings. I check the logs of the 3com NBX, and indeed it receved the full number and passed it straight on to BT successfully, however if I connect * straight to BT , BT don't recognise what I send them... I am quite new to ISDN technologies and am little confused how this can happen, Is there something I am missing here, what else gets passed to the telco from the PRI_CPE end?? Any help would be greatly appreciated Gary ___ 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] A question about queues
Hi there, quick question about queues (B (BWhen calling a queue (which contains eg 4 extensions) it tells me what (Bposition I am in the queue and then plays some music$B!D(Jthat is fine$B!D(J (Bhowever, If there is no-one in the queue , it tells me that im first in line (Band then plays hold music while the phones ring. This is annoying my callers (Bquite a bit . How do I get it so that if I ring the queue, it just puts me (Bstraight through to one of the available 4 phones, and only if all 4 phones (Bare busy (ie on calls) then announce a position in the queue and play music? (B (BFor example (B (BUser 1 dials 7272 $B"*(J goes through to agent 1 (BUser 2 dials 7272 $B"*(J goes through to agent 2 (BUser 3 dials 7272 $B"*(J goes through to agent 3 (BUser 4 dials 7272 $B"*(J goes through to agent 4 (BUser 5 dials 7272 $B"*(J announces message that you are first in line (BUser 6 dials 7272 $B"*(J announces message that you are second in line (B (B (BAny help on this would be greatly appreciated (B (B (B___ (BAsterisk-Users mailing list (BAsterisk-Users@lists.digium.com (Bhttp://lists.digium.com/mailman/listinfo/asterisk-users (BTo UNSUBSCRIBE or update options visit: (B http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] inband DTMF with IAX
Hi, I am currently having a problem where I am making outbound calls via IAX, these calls are then being routed by my provider through a SIP connection to a service providing PSTN access. The problem I have is the the DTMF is being sent inband over the SIP connection, and I am only receiving the DTMF inband on my IAX connection which asterisk is sensibly ignoring as from what I understand with IAX2 the DTMF should be sent out-of-band. I assume the simplest way to fix it is for my provider to put a dtmfmode=inband on the sip.conf entry for their PSTN provider, then presumably their asterisk would see the DTMF and send them out-of-band over the IAX2 channel, however, they are understandably wary of doing this in case it affects any of their other services which currently work perfectly... Is there any way to get Asterisk to listen for inband DTMF from an outbound IAX2 channel so that I can get round this problem in a simple way? (The reason I need to get DTMF on an outbound call is I am trying to set up a 'press 1 to accept the call' system for forwarding calls to mobiles). Thanks in advance, Alex Brett [EMAIL PROTECTED] ___ 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] Cost field in Call Detail Records (cdr)
Hi there I don't know if this utility is available anywhere at the moment but I thought id ask you guys if you know of one What I would like is a way of adding a field to my cdr records (either the Master.csv or a destination mysql table) for cost ! based on some sort of config file (or table) which has a listing of all the tariffs for particular prefixes, ie in the UK, the 0870 prefix is national rate at £0.04p per minute (don't know if that is exact btw !) so I would like a way of adding a field that determines , for example that extension 7201 made a 4 minute call to an 0870 number and therefore that call has cost £0.16p. I can then get my reporting software to pull this additional field into the report Is there anything around that does this sort of thing, open source or otherwise ?? Any help would be greatly appreciated Gary ___ 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] Feedback needed! FindMe/FollowMe Feature Spec.
Hi Brian,
I think this is good that you are doing this; it's a much desired
features. Here are some ideas. Some of which will need to be handled by
some ancillary plug ins.
1. Ability to configure both serial and parallel follow me calls. (see
example)
2. Ability to configure ring seconds for each 'leg' of the followme
circuit
3. Ability to announce call to caller. "This is a follow me call for
'sipuser' push X to accept". User would dial 1 and call would be
connected. If user dials something else (?) or hangs up, call continues
followme patter (caller continues to hear ring or music)
4. Ability to configure if a password is required to connect the call.
Rational here (and can be applied to 2 as well) is this. I use forward
me to many locations. Some of them may not be 'my' phone. Lets say, I'm
in someone elses office I work in frequently. I want the call 1)to be
announced as being for me, and 2) the option to NOT allow anyone but me
accept the call. Seems that the VM password may be good to use here, but
perhaps an override password would be good if there is a separate config
anyway
5. Ability to configure if caller hears rings,music, or a custom
announcement ("Hi this is Brett, please wait while I am located") during
find me operation. Perhaps
6. Ability for the caller to 'break' the followme application. For
example, "Hi this is Brett, please wait while I am located. Press # to
leave a voicemail or 0 to speak to an operator". This is probably best
suited as an option to pass to the application itself: Exten =>
FollowMe(${CONTEXT}${EXTEN},#0) BTW, have a better suggestion than
${CONTEXT}${EXTEN} for multi-tenant configurations?
7. Perhaps have a "ring indefinate" option somewhere too... This may be
a bad reason for billing, security, DOS, etc..
8. Time of day controls!!! Follow me differently on the weekend! Don't
follow me during the workday.
9. Web based follow me setting.
10. Phone based followme setting.
11. A system to override current follow me setting. Two ways of doing
this. One would be a permanent change to follow me settings and
schedule. Ie. I don't care what time of day controls say, use my
"BrettWeekend" schedule. This would be like if the office is closed for
a week for the holidays. Or the other would be a temporary change until
the next timeperiod is met (similar to chaging the temp on a
programmable thermostat). An example here is, I'm leaving work early,
set my Active Schedule to "BrettAfterWork" now. Then when it's time for
"BrettDaytime" to go into effect, it takes back over as usual.
I've implemented follow me systems before, so I've thought about this a
lot. Below is a sample config I could imagine
Example for your config file:
Followme.conf (?) forgive 'pseudo-config' like structures. Please
recommend alternatives if desired.
[sipuser]
; Priority => Extension,Ring Seconds,Announce,Password
; Note: 1 => 7135551212,10,0,1 would probably be confusing and
undesireable
; Ringtype = Music, Ring, or Custom(${soundfile}, ) =1/0
; Ringtype can be at top for default option or in each timeperiod
; Timeperiod = Defined call period in conf file general section or time
range in some format, cron style?
; VMExt = The voicemail extension in the present context
Ringtype => Music
VMExt => 6000
ScheduleName => BrettDaytime
Timeperiod => Daytime
1 => 7135551212,10,1,1
2 => 7135551213,10,1,1
3 => 7135551214,10,0,0
3 => 7135551215,10,1,0
4 => ${VMExt},0,0,0
ScheduleName => BrettOther
Timeperiod => Other
1 => 7135551215,10,1,0
2 => ${VMExt},0,0,0
What do you think?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian D'Arcy
Sent: Tuesday, June 01, 2004 2:15 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Feedback needed! FindMe/FollowMe Feature Spec.
Hello all,
I'm going to tackle learning C this week, and start writing my first *
add-on/contribution; assuming it's actually worthy of contributing once
it's done.. I think I've chosen a hefty project for my first go round
here...
I'd like to get some feedback from everyone on a FindMe/FollowMe spec
I've put together. Before you read on, let me say, I don't want this to
turn into a "it would be cool if it did this.., or that etc..". I'm
writing this to serve a very simple and basic function, and I want it to
do exceedingly well at just that for starters.
Please check out specs below as to how I envision it working within a
dialplan environment, and also, please keep in mind this is being
written to be used in a corporate environment. There are a lot of
others out there with far more * experience than myself, so any
constructive criticism would be most welcome as to the layout and
configuration of the soon to be app_findme
RE: [Asterisk-Users] Asterisk Receptionist manager program.
Ok, Maybe I'm missing something here.. What does "Extension" mean without "Context" also being defined. I don't know what to set in my prefs.conf file for extension... ?? -Brett > >Message: 2 >Date: Thu, 03 Jun 2004 09:27:44 -0700 >From: Kyle Hagan <[EMAIL PROTECTED]> >Organization: Nuvo Technologies >To: [EMAIL PROTECTED] >Subject: Re: [Asterisk-Users] Asterisk Receptionist manager program. >Reply-To: [EMAIL PROTECTED] > > I put a new version up last night. Caller ID shows up on the buttons. >This time IAX is fixed. Works at home and at work through FWD. > >http://www.easyhomenetworks.com/AstRec/ > >Has anyone had anyother bugs popup other than the IAX problem? > >Some people are asking why the screen shot has more buttons than the >alpha version. We are going to get the bugs worked out of the existing >buttons before we add more features. > >Kyle > ___ 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 Receptionist manager program.
Jeremy, Thank you. That is what I mean. And I'm sitting here, looking at the debug window and scratching my head as to HOW he might be using "extension".?? My phone is defined as ACME1000 in sip.conf In extensions.conf I have a: Exten => 1000,1,Dial(SIP/ACME1000) (well, basically) So when I'm at "extension" 1001 (ACME1001 in sip.conf) and I dial 1000. The debug shows the call going to ACME1000. How do you know that "extension" 1000 is ringing?! It seems that you might be making some assumptions on some configuration parameters such as, technology for the extension, context usage, interface naming conventions (sip.conf). I did notice that if I put [EMAIL PROTECTED] for my extension in the prefs.conf file that the voicemail count seems to work. Can you clue me in here as to how to make the other features work. Seems like something I could really use! BTW, I'm a little new to the manager interface, so perhaps I'm just interpreting the messages I see in the debug window. Thanks! -Brett -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeremy Jones Sent: Thursday, June 03, 2004 5:16 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Asterisk Receptionist manager program. I think what he means is this: I can have extension 104 defined in multiple contexts, for instance if I host virtual pbxs for multiple customers on one * box. The syntax of my * conf files requires @ if I want to differentiate between these extensions. If you're using the * box for one business, and you ensure that the same extension is not used in multiple contexts you're ok. But otherwise... Same goes for voicemail -- I can have voicemail box 104 in the voicemail context [great_customer], and another 104 in voicemail context [really_great_customer]. Jeremy -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kyle Hagan Sent: Thursday, June 03, 2004 3:46 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Asterisk Receptionist manager program. This is MY prefs.conf Serverip,192.168.1.40 Port,5038 UID,mark PWD,mysecret MyExt,104 ServerIp = Asterisk Server Port = Port for Manager UID = Manager User PWD = Manager Password MyExt = Extention for Asterisk Manager to Monitor, Transfers, Check VoiceMail etc (my Extention is 104) MyExt is so the person running the application CANT transfer someone elses calls accidentally. It will only effect ext 104's calls. Did I cearify your question? Kyle Brett Nemeroff wrote: >Ok, > >Maybe I'm missing something here.. What does "Extension" mean without >"Context" also being defined. I don't know what to set in my prefs.conf >file for extension... ?? >-Brett > > > > >>Message: 2 >>Date: Thu, 03 Jun 2004 09:27:44 -0700 >>From: Kyle Hagan <[EMAIL PROTECTED]> >>Organization: Nuvo Technologies >>To: [EMAIL PROTECTED] >>Subject: Re: [Asterisk-Users] Asterisk Receptionist manager program. >>Reply-To: [EMAIL PROTECTED] >> >>I put a new version up last night. Caller ID shows up on the buttons. >>This time IAX is fixed. Works at home and at work through FWD. >> >>http://www.easyhomenetworks.com/AstRec/ >> >>Has anyone had anyother bugs popup other than the IAX problem? >> >>Some people are asking why the screen shot has more buttons than the >>alpha version. We are going to get the bugs worked out of the existing >>buttons before we add more features. >> >>Kyle >> >> >> > >___ >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 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] Modem Calls
Scott, I haven't done *any* research into this but.. I'm using a Sipura2000 and I have my laptop modem attached to it. I'm forcing alaw or ulaw encoding on the line. As soon as the modems begin to "train" I get what sounds like "static" instantly "come on". Very unusual. Not sure what the source of it is yet.. Don't laugh, I made my call out on FWD.. :) So I'm not too confident of it working. Let us know if you get modem calls to work. I'm interested as well. Thanks, Brett -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Scott Nelson Sent: Monday, June 07, 2004 1:15 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Modem Calls My office is investigating using an Asterisk PBX and also going to a VOIP provider for our main phone connections, but one of the tricky things is that we need to have outbound and inbound modem calls (fax too). I see a lot of talk about faxes but no mention of modems on this list. I get the impression that modems will be a problem for us. Is that true? ___ 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] Don't want a ring before voice menu
John, If your FXS lines are ringing, it's because your dialplan spells out that the line is to ring. If you don't want it to ring, don't Dial that Zap line. Otherwise If you have an analog phone attached to the same line as the FXO card, you will always hear some amount of ring. Since there is no comprehensive signalling method, the only way Asterisk knows that an analog phone is ringing is to.. Well.. Let it ring and detect that it is happening. -Brett -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Campbell Sent: Tuesday, June 08, 2004 1:55 PM To: asterisk-users Subject: Re: [Asterisk-Users] Don't want a ring before voice menu I should have been clearer in my description of the scenario. What I have is an FXO port connected to an analog line, with 3 FXS extensions. With an incoming call, the phone connected to one of the FXS ports will ring once before kicking in to the voice menu. Eventually, say when the caller presses "0", the call is routed to that extension and the phone rings again the way I've configured it to. Ideally, we would rather not hear the extension ring at all the first time. In short, all incoming calls ring to the extension before the caller gets presented with a voice menu. For now, I've told the receptionist (the person at that extension) to just let it ring. I hope this is somewhat clearer On Tue, 2004-06-08 at 13:52, [EMAIL PROTECTED] wrote: > On Tue, 8 Jun 2004, John Campbell wrote: > > > Hi, > > > > Having searched through the mailing list archives and the wiki, I > > still don't know how to solve the following problem: > > > > Call is received, phone rings once, then the caller gets the voice > > menu. > > > > What I want is for the call not to actually ring, but to go straight > > to the voice menu. > > > > How can I achieve this? > > > > Thanks, > > You are using analog lines? If so, asterisk has no way of knowing the > phone is ringing, until it rings. > > ___ > 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] Call Relaying
The DISA application is traditionally for this functionality. But beware, anything that give outside users the ability to make calls can be abused. -Brett Original message >Date: Sat, 12 Jun 2004 12:20:28 -0400 >From: "Michael Graves" <[EMAIL PROTECTED]> >Subject: [Asterisk-Users] Call Relaying >To: "[EMAIL PROTECTED]" <[EMAIL PROTECTED]> > >Hello All, > >I have a small * server in my home office with several IP phones. The >system is not fully in service yet as I'm still hunting for a cost >effective FXO adapter that I can rely upon for my two primary PSTNs. >That said, I'd like to move it into service for another >application...which brings up a question. > >I'd really like to stop making international calls from my cell phone >when I'm travelling. Can someone point me to an example of extension >logic that accepts an incomming call on a known connection then allows >the caller to access local dialtone to make an international call? > >I have a DID from VoicePulse Connect which I don't really use for much. >I could make that the gateway so that all calls comming in on that DID >have access to outbound dialing. I could also screen the incomming >callerid so that only my or my wife's cell phones get validated. > >Anyone have something comparable that I might look at as a starting >point? > >Thanks, > >Michael > > >-- >Michael Graves [EMAIL PROTECTED] >Sr. Product Specialist www.pixelpower.com >Pixel Power Inc. [EMAIL PROTECTED] > >o713-861-4005 >o800-905-6412 >c713-201-1262 > >"Who do you think you're foolin'?" - Paul Simon > >** Tag(s) inserted by Bandit Tagger98 - http://www.gbar.dtu.dk/~c918704 > > >___ >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] LNP local number portability in Houston (713, 281, 832)
Rich, My organization most likely can assist you with your need. Please contact me off list at [EMAIL PROTECTED] Thanks, Brett -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dr. Rich Murphey Sent: Wednesday, June 16, 2004 3:41 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] LNP local number portability in Houston (713, 281, 832) Are there any VOIP providers that offer Local Number Portability in Houston? Cheers, Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] 911 emergency service and VoIP
Joe, This is highly implementation specific. Perhaps I can give you some pointers to help you out. BTW, if you just happen to be in Texas, I can provide you with a list. "Regular" 911 calls are answered by a PSAP. Voip calls also goto a PSAP, but are handled differently. In fact, in most regions there aren't clear ways of handling these calls as of yet. Here are some pointers. 1. Do NOT call the PSAP. They are very busy, and in general are the WRONG organization to contact. Instead you want the "911 Agency" for the area you are to serve. This is either a "Council Of Governments" or an "Emergency Communications District" depending on when it was formed. For example, here in Houston, the 911 Agency just happens to be the "Harris County ECD". The Houston 911 Agency's website is (coincidential) http://www.911.org You must MUST start with them before you do ANYTHING 911. Certifications are required. http://www.nena.org is a good starting point.. Use "search" 2. If you are going to do 911, you must send LOCATION (ie: address) information to the 911 database. I do this through Intrado http://www.intrado.com through a product called "data exchange" 3. Depending on your connectivity, understand that the 911 agency and PSAP don't care what technology you use to connect to your customer. So if you can provide ANI, you are pretty much good to go. 4. For what it's worth; my traditional VoIP service offering will deliver 911 calls in the indentical manner as my non VoIP calls. If you'd like to talk specifics I can help you but I'd have to request that we take it off list since I feel that it is outside of the scope of the list. You can reach me at [EMAIL PROTECTED] -Brett -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joe Baptista Sent: Wednesday, June 16, 2004 8:35 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] 911 emergency service and VoIP I understand that most VoIP providers allow for 911 calling but that 911 service is not the same as that available to PSTN. >From what I understand a 911 Call Will Go To A General Access Line at the Public Safety Answering Point (PSAP). This is different from the 911 Emergency Response Center where traditional 911 calls go. Does anyone know how I can get information on howto contact the people at the Public Safety Answering Points (PSAPs)? Is there alist somewhere I can reference. thanks joe baptista ___ 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] play gsm files in windows
Does anybody know of a WINDOWS application (preferably freeware) that will simply playback asterisk GSM sound files, I don't want to record them, just playback the ones that are currently there. Any help would be greatly appreciated cheers ___ 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] writing to MYSQL database
Hi there I have been successfully using the asterisk command "MYSQL" to read information from a MySql database, but was wondering if there was any way of WRITING data (ie user input data) to the database ???, looking through the parameters of the MYSQL command it seems as though this function isn't availableis there another application for this??? Cheers ___ 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] 4 port BRI options ?
Hi there I am in the UK.. and am running latest asterisk on FC1 (2.4 kernel). I would like to know what the best option is for a 4 port BRI card. I notice Digium don't provide one.. I have heard the Junghanns do one...but are there others ?? Is the Junghanns card reliable/stable with good sound quality ?? I notice it is very expensive in a per port comparison with the Digium cards hence why I am also looking for alternative cards Your experiences would be greatly appreciated Gary ___ 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] 4 port BRI options ?
Wow that eicon is certainly quite expensive... ive found 2 sellers in the UK so far selling at over £1000 . I think the Junghans comes in at around £600. Is the Eicon that much better ? -Original Message- From: Nardis Dome [mailto:[EMAIL PROTECTED] Sent: 03 June 2005 12:49 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] 4 port BRI options ? Hi, Eicon Diva 4BRI Card and chan_capi. --- "Brett, Gary" <[EMAIL PROTECTED]> wrote: > Hi there > > > I am in the UK.. and am running latest asterisk on > FC1 (2.4 kernel). I would > like to know what the best option is for a 4 port > BRI card. I notice Digium > don't provide one.. I have heard the Junghanns do > one...but are there others > ?? > > Is the Junghanns card reliable/stable with good > sound quality ?? I notice it > is very expensive in a per port comparison with the > Digium cards hence > why I am also looking for alternative cards > > > Your experiences would be greatly appreciated > Gary > ___ > 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 > __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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 ___ 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] defining the zap channel used on inbound analogue calls
Hello all
I am relatively new to asterisk and am sure this will be a simple question
to answer. I have a TDM400p card and I am in the process of creating my dial
plan, however I am a bit stuck on one thing. I have 2 analogue lines (each
obviously with its own DDI) connected to the card; I want to set it up so
that if I dial inbound to the first DDI (e.g. 0208777) it will go to the
IVR and when I ring inbound to the second DDI (e.g. 0208777) I want it
to go directly to the SIP phone internally. Its with the latter I am having
the issue
My problem is this Due to the fact these are analogue lines, I realise
that the DDI is not sent to the TDM400P so I presume the only way for the
dial plan to filter inbound calls is by the Zap Channel it came in on? (In
my case Zap/1 and Zap/2). I have tried the following
--
[globals]
INBOUND=Zap/2
[default]
exten => ${INBOUND},1,Answer
exten => ${INBOUND},2,Background(soundfile),tT
exten => ${INBOUND},3,Hangup
--
I also tried
exten => Zap/2,1,Answer
And
exten => Zap/2-1,1,Answer
And various other combinations all to no avail, is it possible to filter by
the Zap channel used ?, Surely if I want to direct call a phone, I donât
have to go through an IVR everytime ?? (I realise this wouldnât be an issue
with ISDN).
I noticed also in some documentation that you have to use an âsâ for all
analogue traffic, is this the case ?? and if so can you use it in
conjunction with a zap channel definition ??
So in summary, How does the dialplan define the Zap channel used on inbound
analogue calls
Any help would be greatly appreciated
Gary
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