[Asterisk-Users] snom phones and redirect
Hello, does anybody suceesfully setup snom phones with sip firmware with asterisk to redirect call when phone is set to redirect if busy/ or allways redirect ? My console says : chan_sip.c Line 3000 (handle response) Dunno anything about a 302 Moved Temporarily from SIP... regards Marian -- SUNTEQ s. r. o. Bojnicka cesta 35 # Prievidza # 971 04 # Slovak republic Tel: +421-46-5430 754 # Fax: +421-46-5439 144 http://www.sunteq.sk/ A mind is like a parachute... it only works when it's open. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Interest in E1 channel banks?
Very interested! Are you planning for european certification?(expensive!) If so even more interested! Florian Overkamp said: At 18:45 27-2-2003 +1100, you wrote: Our company manufactures an E1 channel bank that is approved for use in Australia (it should also be compatible with Euro standards). It is modular and available in 10, 20 or 30 analog port configurations. Signal monitoring and configuration is via Ethernet. These units are manufactured in low quantities for specific telco requirements. However if there was enough interest, we would be able to manufacture and sell the units at pricing levels under US $2000. So how much interest is out there? /me raises hand (well, we've done our current infra, but it may be a consideration none the less - at these levels of pricing) Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: AGI and fast-entered DTMF codes
Thats fine, didn't expect you to be able to measure it. The no pauses raises a bit of concern. There is supposed to be a set time between keypresses. It must be the phone I was using - which is an internal phone system phone, rather than your average PSTN phone one would have at home. I think my concerns were that if I could cause a problem like that with that particular phone, random users could also cause this. If they are typing fast due to a contest, chances are they already know that they can type faster than speed dial, but they can outrun the telco and have problems. This is something I did back when I tried to win things from the radio stations. At one point I started using a modem that I could tune the pause and tone length till I was getting only about 10% failures. The time to dial was significantly faster. Sorry, I think I explained that a bit wrong. People will receive a code. Then they must call in and enter the code. Once that is successfully verified in the database, they proceed to enter details (for example, date of birth) and then record a short message. The contest is based on the message they enter. Therefore, they will not be in a rush to type in the code faster than others, but my fear was that they might be quick typers and cause a dead line. Again, I used the same (obviously incorrectly configured) phone to call another IVR service that seemed to be able to handle these quick-fire DTMF tones, so I thought that it could be an Asterisk/Zaptel problem, rather than a problem with that particular phone. If it isn't recognised as DTMF, it is sent as audio. There wouldn't be any garbage. I see. Unfortunately, my telecoms experience is very limited (I only really started using Asterisk about 4 weeks ago!), so forgive my understanding of it! Would that audio cause Asterisk problems in that case, if Asterisk were listening out for X DTMF digits, causing it to bomb out of the GET DATA routine? Good, you didn't see that as the begining of a language holy war. For 2 lines you are correct it should be plenty enough. As for the idea of building a multi threaded java server, do you want to tie the system to another single process that could fail and take all lines down with it? We have a app we have been working with, and rather like the idea of the app starting and dieing on a per channel/per call basis so as to make sure the system is always respawning a fresh copy for a call. Our current software has threads that hang occasionally and tie up the phone line till we interactively reset it. This is why I spend so much time with asterisk since it will eliminate that interactive reseting. Excellent point. My background is in the server-side J2EE arena and I recently built a multi-threaded Java server to handle high volume SMS messages routed through 3 network providers. This is a little bit of a paradigm shift from what I was doing, as the calls through Asterisk are interactive and the user knows if something is wrong and can redial. For SMS messages, the user fires and forgets and if no reply is forthcoming, does not know whether that is meant to happen, or the message was lost. This would suggest that, with AGI, the most effective way to go would be a single process that is created and destroyed per call. It does require a trade-off between language functionality and speed of development against the performance profile of that language. Therefore Java might be a bit heavy handed in a large scale environment on that basis. As for the stability of a single server process, monitoring threads can be put in place to ensure a process is in place. However, my preferred way would be to allow the lightweight process that Asterisk calls first to start or restart the server if communication fails. You might have some extra delays, but for high volumes, the extra efficiency might be worth the trade off. Cheers for the help and sorry for going slightly off topic! Alan. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] inbound isdn call
Sounds like the i4l dtmf problem. Assuming you are using i4l, the kernel dtmf detection routines are poor and quite frequently misinterpret speech as dtmf tones. You need to patch asterisk to handle dtmf and i4l not to detect dtmf (or silence). There are a few posts on this list about fixing this issue. Iain --On Thursday, February 27, 2003 12:06 pm +0100 Marian Danisek [EMAIL PROTECTED] wrote: hello, when call is made via asterisk from isdn line to the snom sip phones and caller on isdn line is speaking loudly to the microfone, people on the sip phones didnt hear voice but tones, like dtmf. how can i firuge out this problem ? can echo cancel algorithm ? best regards Marian -- SUNTEQ s. r. o. Bojnicka cesta 35 # Prievidza # 971 04 # Slovak republic Tel: +421-46-5430 754 # Fax: +421-46-5439 144 http://www.sunteq.sk/ A mind is like a parachute... it only works when it's open. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] inbound isdn call
--On Thursday, February 27, 2003 1:54 pm +0100 Marian Danisek [EMAIL PROTECTED] wrote: this mean that i need 2 different patches ? I already found isdn_audio.c and isdn_audio.h patch... this is for i4l. You meat that i need another patch for asterisk ? If you want asterisk to handle dtmf then you need Pauline Middelink's dsp patch - isdn-dsp.txt - which was posted to the list in January. This patch allows asterisk's dsp routines - the same ones used for the zaptel interfaces - to provide dtmf support for the ISDN line. Without it, you will have no dtmf support if you apply the isdn-audio.c patch. Iain ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] logging all console output?
Yes, you can: asterisk -vvvgcn|tee /tmp/log regards Martin On Thu, 27 Feb 2003, Roy Sigurd Karlsbakk wrote: hi can I log all console output while having console access as with asterisk -vvvgc ? -- Roy Sigurd Karlsbakk, Datavaktmester ProntoTV AS - http://www.pronto.tv/ Tel: +47 9801 3356 Computers are like air conditioners. They stop working when you open Windows. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hangup problems...
Hi Guys, Its been almost a week now and I thought I would mention that this seems to have fixed my problem completely. Heh.. of course, the boss is smacking his head against the wall over the fact that he didn't need the T1 card or the channel bank, but.. we would have needed to upgrade in a year or so anyway, so its no big loss. Thanks again for your help! Jim - Original Message - From: Martin Pycko [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, February 24, 2003 2:42 PM Subject: Re: [Asterisk-Users] Hangup problems... It should detect if the party that called in your X100P card hanged up. Then if asterisk hears fast busy it's going to assume that it's the proper time to hang up the channel. But in your case it's not working properly and it's hanging up when there is no fast busy. regards Martin On Mon, 24 Feb 2003, Jim Howarth wrote: Yes, it was in there, I've commented it out and we shall see what happens. Thanks :) Just curious.. do you know what that is for? Jim - Original Message - From: Martin Pycko [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, February 24, 2003 1:57 PM Subject: Re: [Asterisk-Users] Hangup problems... Do you have busydetect=yes in /etc/asterisk/zapata.conf ? If you do then comment it out and see if that helps. regards Martin On Mon, 24 Feb 2003, Jim Howarth wrote: Howdy, We've been trying to get Asterisk to work properly for us and we aren't having much luck when it comes to simply speaking on the phone. We are experiencing hangups don't occur at any specific time. The equipment we use are two X101P's and one T100P. We originally tried with with the USB S100U as well (so channel bank isn't an issue). Sometimes the caller will be completely hung up on and sometimes they will be spit into voicemail. My headset (if it matters) is a Plantronics S10 and the phone is a Bell Venture 3 line phone. Has anyone experienced this? If so, how did you fix it. This is connected to our tech support lines and as you can imagine, customers have a problem with being hung up on - on a regular basis. :D Thanks! Jim ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Intercom and Paging
I forsee one of the following: 1) All phones on same paging system. Uses 1 FXS port or a sound card on the asterisk box. 2) Paging system addressable by groups. Uses 1 FXS port per paging group on asterisk. (Plus 1 or 2 FXS for a 1/2 line phone) 3) Paging system addressable by phone. Uses 1 FXS port per phone dedicated to paging. (Up to 3 FXS per phone, somewhat wasteful.) Since this is my home system and I just want simple paging, I'm going to go for option #1. Obviously a digital phone would make a better solution, but that isn't an option right now. Regards, Jeff On Thu, Feb 27, 2003 at 09:22:06AM -1000, James H. Thompson wrote: Are you going to need two ports on asterisk for each phone or does the intercom connect outside of asterisk? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] snom phones with sip, at asterisk.
How is the behaivor of the snom phones with asterisk? Im thinking of getting 40 snom phones, to manage a voip solution... How's the sound quality? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Intercom and Paging
Jeff - If you make special phone cords that break out the 3rd pair at the wiring closet you can just join those all together in parallel and your paging will be * independant. Bill Jeff Noxon wrote: I forsee one of the following: 1) All phones on same paging system. Uses 1 FXS port or a sound card on the asterisk box. 2) Paging system addressable by groups. Uses 1 FXS port per paging group on asterisk. (Plus 1 or 2 FXS for a 1/2 line phone) 3) Paging system addressable by phone. Uses 1 FXS port per phone dedicated to paging. (Up to 3 FXS per phone, somewhat wasteful.) Since this is my home system and I just want simple paging, I'm going to go for option #1. Obviously a digital phone would make a better solution, but that isn't an option right now. Regards, Jeff On Thu, Feb 27, 2003 at 09:22:06AM -1000, James H. Thompson wrote: Are you going to need two ports on asterisk for each phone or does the intercom connect outside of asterisk? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Intercom and Paging
Yes but I want * to be able to page me, announce queued calls, etc. Basically glorified talking caller ID. ;) On Thu, Feb 27, 2003 at 04:03:09PM -0500, [EMAIL PROTECTED] wrote: Jeff - If you make special phone cords that break out the 3rd pair at the wiring closet you can just join those all together in parallel and your paging will be * independant. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] snom phones with sip, at asterisk.
The SNOM 100's work ok with Asterisk, but the sound quality is poor on the handset, and the speakerphone is unusable. Apparently the latest revision of the SNOM 100 is better, but I haven't used it. The SNOM 200 is supposed to be considerably better, and is only around $50 more. -wade -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Andres Tello Abrego Sent: Thursday, February 27, 2003 3:35 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] snom phones with sip, at asterisk. How is the behaivor of the snom phones with asterisk? Im thinking of getting 40 snom phones, to manage a voip solution... How's the sound quality? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Intercom and Paging
Do any of the SIP phones support intercom/paging? Jim James H. Thompson [EMAIL PROTECTED] - Original Message - From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, February 27, 2003 11:59 AM Subject: Re: [Asterisk-Users] Intercom and Paging Then use 2 lines per phone. You should connect 2 phones together ( the way they want specify ) and hook an oscilloscope to the 3rd pair. It probably uses a simple signal to indicate page-intercom, maybe even DTMF. You then need to rig a voltage convertor from telco batt voltage to whatever they use ( probably 12v or less ) Jeff Noxon wrote: Yes but I want * to be able to page me, announce queued calls, etc. Basically glorified talking caller ID. ;) On Thu, Feb 27, 2003 at 04:03:09PM -0500, [EMAIL PROTECTED] wrote: Jeff - If you make special phone cords that break out the 3rd pair at the wiring closet you can just join those all together in parallel and your paging will be * independant. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Collect Digits for CO Blind Transfer
I have a blind transfer feature available to me from my telephone provider and was wondering if asterisk can take advantage of this so that when a certain extension is called the user is asked for the 11 digit pstn number they wish to call then asterisk flashes the line, dials the transfer codes and hangs up. I have figured out how to do everything except collecting the digits from the user and then using them in the transfer codes. Is it possible to do this with the asterisk dial plan? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Collect Digits for CO Blind Transfer
You can also do that using Background application:
[transfer]
exten = s,1,Background,some-file ;it can be silence
exten = _XXX,1,Flash ;collecting the digits
exten = _XXX,2,SendDTMF,${EXTEN}
exten = _XXX,3,Hangup
[called_context]
exten = 1000,1,Goto,transfer|s|1 ;magical number
regards
Martin
On Thu, 27 Feb 2003, Steven Critchfield wrote:
Not to take away from your prize collection, but for now wouldn't that
be trivial to do either as a patternmatch with assignment to a variable,
or slightly less trivialy as a agi app that already has access to the
getdata function?
On Thu, 2003-02-27 at 17:42, Mark Spencer wrote:
Sounds like we need an app_getdata (front end to the getdata function).
I'll do it for any item in my thinkgeek wishlist (you can search by
[EMAIL PROTECTED] or by my name).
Mark
On Thu, 27 Feb 2003, Ben Clark wrote:
I have a blind transfer feature available to me from my telephone
provider and was wondering if asterisk can take advantage of this so
that when a certain extension is called the user is asked for the 11
digit pstn number they wish to call then asterisk flashes the line,
dials the transfer codes and hangs up. I have figured out how to do
everything except collecting the digits from the user and then using
them in the transfer codes. Is it possible to do this with the
asterisk dial plan?
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--
Steven Critchfield [EMAIL PROTECTED]
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