Re: [asterisk-users] misdn and incoming fax detection
hmmm.. no ideas?! :-| tom Thomas Artner wrote: Hi! At the moment i am using a digium tdm400 card for my analog phone lines. The zaptel driver supports fax detection, so incoming faxes are redirected to the fax extension automatically. This works without problems with asterisk 1.2. But now I would like to switch to ISDN (mISDN) and asterisk 1.4. I have a ISDN Card which supports the mISDN channel. Everything is working fine, but I dont know how I can do such incoming fax detection as I did it with the tdm400 card. :-( Is the only way using nvFaxDetect? Its terrible to get it run with asterisk 1.4. As well http://www.newmantelecom.com/asterisk/faxdetect/ seems not to be available anymore. Has anyone some ideas how I can do such incoming fax detection with mISDN and asterisk 1.4 ? thx, Tom ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] misdn and incoming fax detection
Hi! At the moment i am using a digium tdm400 card for my analog phone lines. The zaptel driver supports fax detection, so incoming faxes are redirected to the fax extension automatically. This works without problems with asterisk 1.2. But now I would like to switch to ISDN (mISDN) and asterisk 1.4. I have a ISDN Card which supports the mISDN channel. Everything is working fine, but I dont know how I can do such incoming fax detection as I did it with the tdm400 card. :-( Is the only way using nvFaxDetect? Its terrible to get it run with asterisk 1.4. As well http://www.newmantelecom.com/asterisk/faxdetect/ seems not to be available anymore. Has anyone some ideas how I can do such incoming fax detection with mISDN and asterisk 1.4 ? thx, Tom ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk and fax machine
Hi! Either the fax machine or the asterisk box has to pick up the call to know whether it is a fax or not. My solution is that I let asterisk pick up every call, and if it is a fax, then the call is forwarded to a fax-machine. If its a voice call, the call is forwarded to the phones. [incoming] exten = s,1,Answer() ;automatic answer for fax recognition exten = s,2,Wait(3);prevents ringing when it is a fax exten = s,3,Dial(Sip/21Sip/22Sip/25Sip/26,45,t) ;ring phones exten = s,4,Hangup ;hangup after 45 secondes ;is it a fax? then take it here! exten = fax,1,Dial(Zap/1) But this solution implies that asterisk picks up every call immediately. So the caller has to pay for the call before he can talk to you. tom aslay-pinwee wrote: Hi, I need to share my PSTN line with my Digium card together with my FAX machine. If fax coming in, will asterisk pick up the call or my fax machine pick up the call. How do I make asterisk not to answer the incoming fax and let my fax machine receive the fax. Similarly, how do I make my fax machine not to answer any voice call and let my asterisk answer.. Regards ASLAY ___ --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 ___ --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] a/b door phone
Hi! I would like to connect a door phone to my asterisk server. I decided to do that with an analog (a/b) doorphone and a sipura box. Can anyone give me a recommendation for a door phone with good voice quality? I am from Austria/Europe and I looked at products from Rocom and from Auerswald. Does anyone use one of these door phones and can me tell his/her opinion on that?! thx in advance, tom ___ --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] 64 analog phones
It depends on the actual given environment, but you could also think about using Linksys' PAP2 adapter! mike wrote: Dear list which hardware solution would you suggest for connecting 60 analog phones to asterisk ? thank you very much .mike ___ --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 ___ --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] Unknown RTP codec 100 received
Hi! I have an analog fax machine connected to a sipura ATA which is connected to my asterisk box. In my asterisk box I have a digium card for a connection to the public telephone network (analog). On this digium card is also a us robotics sportster modem (analog) connected. My problem: I can send faxes from the analog faxmachine to the us robotics modem, but when I try to send a fax to someone else (via the public phone network) it fails and I get the following error message: Unknown RTP codec 100 received. Does anyone have an idea why faxing to the analog modem works perfectly, and why it fails if i try to send one outside ? thx in advance, tom ___ --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] E61
Hi! I can't find the link anymore where it was a statement from the nokia support that they are working on a STUN implementation. A firmwareupdate (with STUN support) will be available in fall 2006. tom Andreas Sikkema wrote: Anyone here use the Nokia E61 ? I am looking to invest in a wifi phone and I want to get the best. Is it good as far as reception ? That is of most importance to me. Thanks. I've tried it in the last couple of days. The biggest issue for me ist that it HAS to be on the same side of a NAT as the server it talks to (asterisk, ser, etc). If it is on the private side of a NAT and the server is on the public side, it doesn't work. I've read something on the Nokia forums that Nokia is aware of the problem and it will be solved. My problem is that they want to solve this using STUN etc, while I would prefer they also wouldn't have the software care if it is on the inside of a NAT like most other CPE's so our platform can take care of things. ___ --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] E61
here is the link (german): http://www.my-s60.com/de/news/news/newswann_kann_die_e_serie_nat_traversal/back/105/cHash/bac3645f5e/index.html STUN will come in fall 2006 TURN and ICE in 2007 Thomas Artner wrote: Hi! I can't find the link anymore where it was a statement from the nokia support that they are working on a STUN implementation. A firmwareupdate (with STUN support) will be available in fall 2006. tom Andreas Sikkema wrote: Anyone here use the Nokia E61 ? I am looking to invest in a wifi phone and I want to get the best. Is it good as far as reception ? That is of most importance to me. Thanks. I've tried it in the last couple of days. The biggest issue for me ist that it HAS to be on the same side of a NAT as the server it talks to (asterisk, ser, etc). If it is on the private side of a NAT and the server is on the public side, it doesn't work. I've read something on the Nokia forums that Nokia is aware of the problem and it will be solved. My problem is that they want to solve this using STUN etc, while I would prefer they also wouldn't have the software care if it is on the inside of a NAT like most other CPE's so our platform can take care of things. ___ --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 ___ --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] Idiot questions
joea, j4computers wrote: As a complete newcomer to Asterisk, Digium and PBX, I have several questions. But I'll start with this. To setup a simple system with only a couple of POTS lines, I gather I will need a TDM400 board with FXO and/or FXS modules. So, a TDM400 card will support up to two analog (POTS) lines? a tdm400 card has 4 slots. each of these slots can be assembled with a FXS or FXO module. So you can handle 4 FXO lines, or 4 FXS, or 2FXO and 2 FXS but its recommended to use only one tdm400 card per computer. joea ___ --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 ___ --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] strange behaviour of a zaptel device
Hi! I am working hard on getting a useful attented transfer. (The built-in atxfer feature isnt useful - because of calls getting lost - has been discussed a few months ago) I have all my analog phones on sipura boxes. With the flash hook i can do such attended transfers without problems now. But, the asterisk box is connected to a POTS line via a digium card. And here I have a strange behaviour: .) a call comes in (via the digium card) .) Person A takes the call (on a sipura pap or spa device) .) Person A presses the flash button and dials an other extension for Person B .) Person B hooks the phone off (sipura pap or spa), talks a few words with Person A .) Person A hooks the phone on. .) the incoming call is transferred to Person B, BUT the caller on the incoming call cant hear Person B, while Person B can hear the incoming caller! Does anyone have any hints for me ? Asterisk at log level 10 doesnt show anything... :-( thx in advance, Tom ___ --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] strange behaviour of a zaptel device SOLVED
Thomas Artner wrote: Hi! I am working hard on getting a useful attented transfer. (The built-in atxfer feature isnt useful - because of calls getting lost - has been discussed a few months ago) I have all my analog phones on sipura boxes. With the flash hook i can do such attended transfers without problems now. But, the asterisk box is connected to a POTS line via a digium card. And here I have a strange behaviour: .) a call comes in (via the digium card) .) Person A takes the call (on a sipura pap or spa device) .) Person A presses the flash button and dials an other extension for Person B .) Person B hooks the phone off (sipura pap or spa), talks a few words with Person A .) Person A hooks the phone on. .) the incoming call is transferred to Person B, BUT the caller on the incoming call cant hear Person B, while Person B can hear the incoming caller! in sip.conf: canreinvite = no solved this issue Does anyone have any hints for me ? Asterisk at log level 10 doesnt show anything... :-( thx in advance, Tom ___ --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 ___ --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] Sending Email From A Dial Plan
Damien Gabrielson wrote:
Hi,
I'm looking for a simple way to send email from a dial plan. I have
searched around quite a bit looking for a solution for this and I'm
surprised that I haven't found anything useful yet other than using the
System() application. I would like to be able to change the subject
dynamically based on ${EXTEN} and the body is not important. I was
hoping to have a one line command from the System() application without
having to write a script or any other dependency. Has anyone implemented
anything like this?
How about
System(echo message text | mail -s ${EXTEN} [EMAIL PROTECTED]) ?
Why is it a problem to use the system command? You can set the subject
in dependence of ${EXTEN}!
Thanks,
Damien
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[asterisk-users] Attended Transfer call return with asterisk + sipura spa2002
Hi! I have connected my analog phones to an asterisk box with sipura spa2002 devices. I can do an attended transfer by taking the call which should be transferred, pressing the flash button, dialing the number to which the call should be transferred and now i can hang up or talk to the person who receives the transferred call. Thats working perfectly. But if the other person (the person who gets the transferred call) isnt on his place and doesnt take the call, the call gets disconnected after about a minute. Does anyone have an idea how i could make that the call gets transferred back (to the person who initially did the transfer) automatically? thx, Tom ___ --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 check the status of a channel
Alexander Lopez wrote: This might be what you're seeking; http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+ChanIsAvail If the phone rings, then the channel IS available. The solution is to disable call waiting on the SIP device. The s option needs to be used: s - Consider the channel unavailable if the channel is in use at all Thats it! Thx a lot :-) tom ___ --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 ___ --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] how to check the status of a channel
Hi! I have two extensions (25 and 26, and so two phones) for one person in an office. I can dial 25 or 26 and always both extensions are ringing. Thats okay! exten = 25,1,Dial(Sip/25Sip/26) exten = 26,1,Dial(Sip/25Sip/26) The problem with this solution is, if the person is talking on one phone and 25 or 26 is called from anywhere, the other phone is ringing. But I would like a busy signal if the person is talking on one of these two phones. How could I do that in the dialplan? I couldn't find something to check whether one of these two channels is busy or not. Any suggestions for me? thx, Thomas ___ --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] tdm400p card for sell (4xFXS)
Hi! I've one card over from my last asterisk project. The card is about 3 months old, a copy from the invoice for warranty is available. Location: Vienna, Austria. If anyone is interested - send me a private mail. cheers, Tom (i hope this mail is okay for this list) ___ --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] Question about the zaptel-1.2.5-patch
Am Wednesday 26 April 2006 20:43 schrieb Wai Wu: If I download zaptel-1.2.5, do I still have to apply the zaptel-1.2.5-patch? no. ___ --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 ___ --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] annoying noise on analog phones on tdm400p
hmm.. does really nobody had such an issue before? Thomas Artner wrote: Hi! I am using asterisk with two tdm400p cards. Sometimes (one call out of ten), when a call comes in and is taken, there is some terrible noise for a short time in the line (for about a second). Both partys can hear the noise. And sometimes the call has to be hung up, because the noise doesn't disappear. Has anyone any idea where the problem could be? cheers, tom ___ --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 ___ --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] annoying noise on analog phones on tdm400p
Hi! I am using asterisk with two tdm400p cards. Sometimes (one call out of ten), when a call comes in and is taken, there is some terrible noise for a short time in the line (for about a second). Both partys can hear the noise. And sometimes the call has to be hung up, because the noise doesn't disappear. Has anyone any idea where the problem could be? cheers, tom ___ --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] attended transfer issue
Hi! I decided to open an issue about this case in the mantis database! I am not very familiar with the bug/issue tracking procedure at the asterisk project, but I think i can make it. Is there something that would speak against it? cheers, tom Thomas Artner wrote: Hi! A few months ago I needed some help for the following issue: .) a call comes in .) Person A takes the call and does an attended transfer to Person B .) Person A hangs up the phone without waiting for Person B taking the call .) the caller get lost at this point !! At this point the attended transfer should go into a blind transfer. The phone of Person B should still be ringing and the caller shouldnt get lost. I think this is the most usual behaviour of a call transfer also on the cheapest systems on the market. Why doesnt this work well with asterisk? Will there be a solution for that in the near future? I am thankful for any kind of help! thx, Tom ___ --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 ___ --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] attended transfer issue
here's the reported issue: http://bugs.digium.com/view.php?id=6973 cheers, tom Thomas Artner wrote: Hi! I decided to open an issue about this case in the mantis database! I am not very familiar with the bug/issue tracking procedure at the asterisk project, but I think i can make it. Is there something that would speak against it? cheers, tom Thomas Artner wrote: Hi! A few months ago I needed some help for the following issue: .) a call comes in .) Person A takes the call and does an attended transfer to Person B .) Person A hangs up the phone without waiting for Person B taking the call .) the caller get lost at this point !! At this point the attended transfer should go into a blind transfer. The phone of Person B should still be ringing and the caller shouldnt get lost. I think this is the most usual behaviour of a call transfer also on the cheapest systems on the market. Why doesnt this work well with asterisk? Will there be a solution for that in the near future? I am thankful for any kind of help! thx, Tom ___ --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 ___ --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 ___ --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] asterisk 1.2.7.1 and app_rxfax
Hi! After upgrading to asterisk 1.2.7.1 (from 1.2.4) it seems that app_rxfax doesnt work any more. I've installed spandsp-0.0.2pre25 (the same problem with spandsp-0.0.3pre6.tgz ) app_rxfax.c, app_txfax.c and made the Makefile patch from the same directory. When starting asterisk I always get the follwing error message: [app_rxfax.so]Apr 14 18:54:20 WARNING[7223]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/app_rxfax.so: undefined symbol: t30_get_far_ident Apr 14 18:54:20 WARNING[7223]: loader.c:554 load_modules: Loading module app_rxfax.so failed! Does anyone have any idea how to fix that? cheers, Tom ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk 1.2.7.1 and app_rxfax
Rob Terhaar wrote: did you try to recompile the plugin? yes, of course... On 4/14/06, Thomas Artner [EMAIL PROTECTED] wrote: Hi! After upgrading to asterisk 1.2.7.1 (from 1.2.4) it seems that app_rxfax doesnt work any more. I've installed spandsp-0.0.2pre25 (the same problem with spandsp-0.0.3pre6.tgz ) app_rxfax.c, app_txfax.c and made the Makefile patch from the same directory. When starting asterisk I always get the follwing error message: [app_rxfax.so]Apr 14 18:54:20 WARNING[7223]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/app_rxfax.so: undefined symbol: t30_get_far_ident Apr 14 18:54:20 WARNING[7223]: loader.c:554 load_modules: Loading module app_rxfax.so failed! Does anyone have any idea how to fix that? cheers, Tom ___ --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 ___ --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 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk 1.2.7.1 and app_rxfax
after a few hours of debugging it works now... I got some version mixes of spandsp on my system... sorry for the spam tom Thomas Artner wrote: Hi! After upgrading to asterisk 1.2.7.1 (from 1.2.4) it seems that app_rxfax doesnt work any more. I've installed spandsp-0.0.2pre25 (the same problem with spandsp-0.0.3pre6.tgz ) app_rxfax.c, app_txfax.c and made the Makefile patch from the same directory. When starting asterisk I always get the follwing error message: [app_rxfax.so]Apr 14 18:54:20 WARNING[7223]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/app_rxfax.so: undefined symbol: t30_get_far_ident Apr 14 18:54:20 WARNING[7223]: loader.c:554 load_modules: Loading module app_rxfax.so failed! Does anyone have any idea how to fix that? cheers, Tom ___ --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 ___ --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] attended transfer issue
Hi! A few months ago I needed some help for the following issue: .) a call comes in .) Person A takes the call and does an attended transfer to Person B .) Person A hangs up the phone without waiting for Person B taking the call .) the caller get lost at this point !! At this point the attended transfer should go into a blind transfer. The phone of Person B should still be ringing and the caller shouldnt get lost. I think this is the most usual behaviour of a call transfer also on the cheapest systems on the market. Why doesnt this work well with asterisk? Will there be a solution for that in the near future? I am thankful for any kind of help! thx, Tom ___ --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] attended transfer issue
Michael Collins wrote: A few months ago I needed some help for the following issue: .) a call comes in .) Person A takes the call and does an attended transfer to Person B .) Person A hangs up the phone without waiting for Person B taking the call .) the caller get lost at this point !! At this point the attended transfer should go into a blind transfer. The phone of Person B should still be ringing and the caller shouldnt get lost. I think this is the most usual behaviour of a call transfer also on the cheapest systems on the market. Could you remind us of what kinds of phones you are using, and whether you're using SIP, Zap or something else? i am using analog phones on digium cards (zaptel). Thanks! -MC ___ --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 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk 1.2.7.1 and app_rxfax
Melcon Moraes wrote: So, what version of spandsp are using afterall? i am using spandsp-0.0.2pre25 now. In the 0.0.3 package, there is no app_rxfax.c and no app_txfax.c. No idea why thats missing there. tom []'s MM -Original Message- From: Thomas Artner [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Cc: Sent: Fri, 14 Apr 2006 19:50:47 +0200 Delivered: Fri, 14 Apr 2006 11:52:45 Subject:[Asterisk-Users] asterisk 1.2.7.1 and app_rxfax after a few hours of debugging it works now... I got some version mixes of spandsp on my system... sorry for the spam tom Thomas Artner wrote: Hi! After upgrading to asterisk 1.2.7.1 (from 1.2.4) it seems that app_rxfax doesnt work any more. I've installed spandsp-0.0.2pre25 (the same problem with spandsp-0.0.3pre6.tgz ) app_rxfax.c, app_txfax.c and made the Makefile patch from the same directory. When starting asterisk I always get the follwing error message: [app_rxfax.so]Apr 14 18:54:20 WARNING[7223]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/app_rxfax.so: undefined symbol: t30_get_far_ident Apr 14 18:54:20 WARNING[7223]: loader.c:554 load_modules: Loading module app_rxfax.so failed! Does anyone have any idea how to fix that? cheers, Tom ___ --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 ___ --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 E-mail classificado pelo Identificador de Spam Inteligente Terra. Para alterar a categoria classificada, visite http://mail.terra.com.br/protected_email/imail/imail.cgi?+_u=levelz_l=1,1145037166.47636.6654.arrino.terra.com.br,4381,Des15,Des15 --Original Message Ends-- ___ --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] Re: Attended Transfer - transfer timeout, how to change?
Tomislav Parčina wrote: In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... you are using the attended transfer feature.. ist it already possible to hang up before the other person lifts the handset without loosing the caller when you are doing an attendet transfer? (person A takes an incoming call, person A would like to do an attended transfer to person B, person A hangs up the phone BEFORE person B takes the transfered call -- does the incoming call get lost?) this was an issue in 1.2.4, I'd like to know whether its fixed in 1.2.5. You shouldn't hang up. You should use disconnect = #0 from features.conf+ Yes - but thats not really comfortable :-( -- Tomislav Parcina tparcina#lama.hr ___ --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 ___ --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] Attended Transfer - transfer timeout, how to change?
Am Tuesday 14 March 2006 18:38 schrieb Barry Flanagan: Hi, We are trying to use attended transfer with Asterisk 1.2.5, but when we do the transfer and dial the new number, it times out after 3 rings and then the callee is put back to the original agent. Where can I adjust the timeout which applies to the number we are transferring to? I have changed the extension for this number to timeout at 60 seconds, but that seems to make no difference. try to adjust the parkingtime parameter in features.conf. you are using the attended transfer feature.. ist it already possible to hang up before the other person lifts the handset without loosing the caller when you are doing an attendet transfer? (person A takes an incoming call, person A would like to do an attended transfer to person B, person A hangs up the phone BEFORE person B takes the transfered call -- does the incoming call get lost?) this was an issue in 1.2.4, I'd like to know whether its fixed in 1.2.5. greets, Tom ___ --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] fax receive using TDM400P
Am Monday 27 February 2006 23:15 schrieb Anton Krall: Guys.. I just thought of something.. Anybody who is sucessfuly receviing faxes using spandsp and running Fedora Core 3? What are you running? Debian stable - and it works perfectly. |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Anton Krall |Sent: Saturday, February 25, 2006 6:03 AM |To: 'Asterisk Users Mailing List - Non-Commercial Discussion' |Subject: RE: [Asterisk-Users] fax receive using TDM400P | |That's what I imagined. I read somewhere that echocancel kills |faxes.. Also, I guess hardward cards with echo cancel modules |are a nono :)? | ||-Original Message- ||From: [EMAIL PROTECTED] ||[mailto:[EMAIL PROTECTED] On Behalf Of Lee ||Howard ||Sent: Friday, February 24, 2006 10:45 AM ||To: Asterisk Users Mailing List - Non-Commercial Discussion ||Subject: Re: [Asterisk-Users] fax receive using TDM400P || ||Anton Krall wrote: ||Any modification made to zapata as far as echo and gains? || ||As a rule, don't let anything manipulate the audio at all... ||even echo cancellation. That said, I have seen situations where gain ||had to be increased. || ||Should echocancel be on or off? || ||Off, most definitely off. I can't imagine an echo cancellor being ||capable of knowing what is echo and what isn't echo in a fax call. || ||Lee. ___ --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 ___ --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] fax receive using TDM400P
Anton Krall wrote: Ok 1 for Debian, any Fedoras Core 3 out there? I think it doesn't depend on the linux distribution whether it works or not. It's rather an hardware issue. |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Thomas Artner |Sent: Monday, February 27, 2006 4:57 PM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] fax receive using TDM400P | |Am Monday 27 February 2006 23:15 schrieb Anton Krall: | Guys.. I just thought of something.. Anybody who is sucessfuly | receviing faxes using spandsp and running Fedora Core 3? | What are you running? | |Debian stable - and it works perfectly. | | ___ --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 ___ --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] fax receive using TDM400P
Am Saturday 25 February 2006 23:49 schrieb Anton Krall:
Whats mpack tom?
a command line tool for easily sending emails with attachments.
I use sendEmail..
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Thomas Artner
|Sent: Saturday, February 25, 2006 4:25 PM
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: Re: [Asterisk-Users] fax receive using TDM400P
|
|Am Saturday 25 February 2006 22:59 schrieb Anton Krall:
| I cant get faxes right now with tdm, something is wrong but,
|
|what do I
|
| need to have in order to convert from tiff to pdf?
|
| I have the mailfax script that invokes tif2ps and ps2pdf but pages
| come out blank..
|
|I do the following:
|
|exten = fax,1,Set(FAXFILE=/var/spool/asterisk-fax/${UNIQUEID})
|exten = fax,2,rxfax(${FAXFILE})
|exten = fax,3,system(tiff2pdf ${FAXFILE} ${FAXFILE}.pdf)
|exten = fax,4,system(mpack -s received Fax -c
|application/octet-stream ${FAXFILE}.pdf [EMAIL PROTECTED])
|
|
|
|
|tom
|
| |-Original Message-
| |From: [EMAIL PROTECTED]
| |[mailto:[EMAIL PROTECTED] On Behalf Of
| |Thomas Artner
| |Sent: Saturday, February 25, 2006 12:47 PM
| |To: Asterisk Users Mailing List - Non-Commercial Discussion
| |Subject: Re: [Asterisk-Users] fax receive using TDM400P
| |
| |Am Saturday 25 February 2006 19:38 schrieb Steve Underwood:
| | Cosmin Prund wrote:
| | I've noticed some other odd thing with rxfax. In my case I can
| | receive faxes (using TDM400P) just fine. I can only see
| |
| |those faxes
| |
| | using Windows XP's Fax and Picture thingy, other
| |
| |applications are having trouble.
| |
| | Also printing those faxes is a bit odd: the preview is
| |
| |just fine but
| |
| | I always need to specify landscape printing for
| |
| |portrait faxes.
| |
| | If I print an portrait fax using potrait setting the fax is
| | actually printed landscape, shrinked on it's vertical
| |
| |dimension and
| |
| | widend on it's horizantal dimension. Really funny! I
|
|don't know if
|
| | this is a problem with the viewer application or with the
| |
| |tiff file itself...
| |
| | Its the viewers. A large number of TIFF viewers are badly
| |
| |broken. Some
| |
| | only show the first page. Some do not obey the standard/fine
| | resolution things properly, and get things very squashed.
| |
| |i think the better way is to convert the tiff to pdf before
| |sending the file to the enduser!
| |
| |
| |
| |tom
| |
| | Steve
| |
| | ___
| | --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
| |
| |___
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| |
| |Asterisk-Users mailing list
| |To UNSUBSCRIBE or update options visit:
| | http://lists.digium.com/mailman/listinfo/asterisk-users
|
| ___
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|
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|http://lists.digium.com/mailman/listinfo/asterisk-users
|
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| http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [Asterisk-Users] fax receive using TDM400P
Am Saturday 25 February 2006 19:38 schrieb Steve Underwood: Cosmin Prund wrote: I've noticed some other odd thing with rxfax. In my case I can receive faxes (using TDM400P) just fine. I can only see those faxes using Windows XP's Fax and Picture thingy, other applications are having trouble. Also printing those faxes is a bit odd: the preview is just fine but I always need to specify landscape printing for portrait faxes. If I print an portrait fax using potrait setting the fax is actually printed landscape, shrinked on it's vertical dimension and widend on it's horizantal dimension. Really funny! I don't know if this is a problem with the viewer application or with the tiff file itself... Its the viewers. A large number of TIFF viewers are badly broken. Some only show the first page. Some do not obey the standard/fine resolution things properly, and get things very squashed. i think the better way is to convert the tiff to pdf before sending the file to the enduser! tom Steve ___ --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 ___ --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] fax receive using TDM400P
Am Saturday 25 February 2006 22:59 schrieb Anton Krall:
I cant get faxes right now with tdm, something is wrong but, what do I need
to have in order to convert from tiff to pdf?
I have the mailfax script that invokes tif2ps and ps2pdf but pages come out
blank..
I do the following:
exten = fax,1,Set(FAXFILE=/var/spool/asterisk-fax/${UNIQUEID})
exten = fax,2,rxfax(${FAXFILE})
exten = fax,3,system(tiff2pdf ${FAXFILE} ${FAXFILE}.pdf)
exten = fax,4,system(mpack -s received Fax -c application/octet-stream
${FAXFILE}.pdf [EMAIL PROTECTED])
tom
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Thomas Artner
|Sent: Saturday, February 25, 2006 12:47 PM
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: Re: [Asterisk-Users] fax receive using TDM400P
|
|Am Saturday 25 February 2006 19:38 schrieb Steve Underwood:
| Cosmin Prund wrote:
| I've noticed some other odd thing with rxfax. In my case I can
| receive faxes (using TDM400P) just fine. I can only see
|
|those faxes
|
| using Windows XP's Fax and Picture thingy, other
|
|applications are having trouble.
|
| Also printing those faxes is a bit odd: the preview is
|
|just fine but
|
| I always need to specify landscape printing for
|
|portrait faxes.
|
| If I print an portrait fax using potrait setting the fax is
| actually printed landscape, shrinked on it's vertical
|
|dimension and
|
| widend on it's horizantal dimension. Really funny! I don't know if
| this is a problem with the viewer application or with the
|
|tiff file itself...
|
| Its the viewers. A large number of TIFF viewers are badly
|
|broken. Some
|
| only show the first page. Some do not obey the standard/fine
| resolution things properly, and get things very squashed.
|
|i think the better way is to convert the tiff to pdf before
|sending the file to the enduser!
|
|
|
|tom
|
| Steve
|
| ___
| --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
|
|___
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|
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|To UNSUBSCRIBE or update options visit:
| http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [Asterisk-Users] fax receive using TDM400P
Hi! I am using tdm400 cards for receiving faxes. It worked quite out of the box. I installed spandsp for the rxfax application only. I use it as phone/fax switch: All incoming calls are answered automatically to listen whether its a fax or not. If it is a fax, the call is forwarded to the buil-in fax extension, otherwise the analog phones (all on tdm400) rings. It works without problems. Its for a small company (about a few faxes per hour) Tom Am Freitag, 24. Februar 2006 07:10 schrieb Anton Krall: Guys. Ive been testing how to receive faxes using TDM400P cards and so far, after playing with gains, echocancell and echotraining on zapata.conf.. Ive ha dno luck, faxes come in as garbage or broken or with blank lines. Anybody has successfully done this? Any tips.. Also I have some ideas: 1. Is it really possible to get fxes on a fax machine using ATAs like the sipura 2002? Even using ulaw and pass-thru, is it possible? 2. Since the faxes is coming from PSTN thru the card, I guess asterisk will always stay in the middle right? No way around this. 3. Im also trying to receive faxes usign a TE110P card with spandsp, unicall and E1 R2MFC, no luck also, some stuff, garbage and broken faxes. Anybody done this sucessfuly? Hope anybody can share their thoughts and insight on this. ___ --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 -- Thomas Artner ___ --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] fax receive using TDM400P
Am Friday 24 February 2006 16:48 schrieb Anton Krall: Any modification made to zapata as far as echo and gains? Should echocancel be on or off? i have echocancel switched on, faxdetect is on, rx- and txgain is not used. (commented out). my var/log/messages says: Found a Wildcard TDM: Wildcard TDM400P REV I (4 modules) ... Zaptel Version: 1.2.4 Echo Canceller: KB1 maybe it depends on different hardware revisions? i don't know... tom |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Thomas Artner |Sent: Friday, February 24, 2006 8:25 AM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] fax receive using TDM400P | |Hi! | |I am using tdm400 cards for receiving faxes. It worked quite |out of the box. I installed spandsp for the rxfax application only. | |I use it as phone/fax switch: |All incoming calls are answered automatically to listen |whether its a fax or not. If it is a fax, the call is |forwarded to the buil-in fax extension, otherwise the analog |phones (all on tdm400) rings. | |It works without problems. Its for a small company (about a |few faxes per |hour) | | |Tom | |Am Freitag, 24. Februar 2006 07:10 schrieb Anton Krall: | Guys. | | Ive been testing how to receive faxes using TDM400P cards | |and so far, after | | playing with gains, echocancell and echotraining on | |zapata.conf.. Ive ha | | dno luck, faxes come in as garbage or broken or with blank lines. | | Anybody has successfully done this? Any tips.. Also I have | |some ideas: | 1. Is it really possible to get fxes on a fax machine using | |ATAs like the | | sipura 2002? Even using ulaw and pass-thru, is it possible? | | 2. Since the faxes is coming from PSTN thru the card, I | |guess asterisk will | | always stay in the middle right? No way around this. | | 3. Im also trying to receive faxes usign a TE110P card with spandsp, | unicall and E1 R2MFC, no luck also, some stuff, garbage and | |broken faxes. | | Anybody done this sucessfuly? | | Hope anybody can share their thoughts and insight on this. | | ___ | --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 | |-- |Thomas Artner |___ |--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 ___ --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 ___ --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] fax receive using TDM400P
Am Friday 24 February 2006 21:17 schrieb yrving rivas: Thomas: does it work in your case? Do anybody have the fax working w/tdm? Yes, as I wrote before, receiving faxes with a tdm400p card works perfectly! Thomas Artner [EMAIL PROTECTED] escribió: Am Friday 24 February 2006 16:48 schrieb Anton Krall: Any modification made to zapata as far as echo and gains? Should echocancel be on or off? i have echocancel switched on, faxdetect is on, rx- and txgain is not used. (commented out). my var/log/messages says: Found a Wildcard TDM: Wildcard TDM400P REV I (4 modules) ... Zaptel Version: 1.2.4 Echo Canceller: KB1 maybe it depends on different hardware revisions? i don't know... tom |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Thomas Artner |Sent: Friday, February 24, 2006 8:25 AM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] fax receive using TDM400P | |Hi! | |I am using tdm400 cards for receiving faxes. It worked quite |out of the box. I installed spandsp for the rxfax application only. | |I use it as phone/fax switch: |All incoming calls are answered automatically to listen |whether its a fax or not. If it is a fax, the call is |forwarded to the buil-in fax extension, otherwise the analog |phones (all on tdm400) rings. | |It works without problems. Its for a small company (about a |few faxes per |hour) | | |Tom | |Am Freitag, 24. Februar 2006 07:10 schrieb Anton Krall: | Guys. | | Ive been testing how to receive faxes using TDM400P cards | |and so far, after | | playing with gains, echocancell and echotraining on | |zapata.conf.. Ive ha | | dno luck, faxes come in as garbage or broken or with blank lines. | | Anybody has successfully done this? Any tips.. Also I have | |some ideas: | 1. Is it really possible to get fxes on a fax machine using | |ATAs like the | | sipura 2002? Even using ulaw and pass-thru, is it possible? | | 2. Since the faxes is coming from PSTN thru the card, I | |guess asterisk will | | always stay in the middle right? No way around this. | | 3. Im also trying to receive faxes usign a TE110P card with spandsp, | unicall and E1 R2MFC, no luck also, some stuff, garbage and | |broken faxes. | | Anybody done this sucessfuly? | | Hope anybody can share their thoughts and insight on this. | | ___ | --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 | |-- |Thomas Artner |___ |--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 ___ --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 ___ --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 - Do You Yahoo!? La mejor conexión a Internet y 2GB extra a tu correo por $100 al mes. http://net.yahoo.com.mx ___ --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] Multiple TDM400P's in a single machine
Am Tuesday 21 February 2006 00:24 schrieb Marc Archer: Hi All, Can someone give me a definite answer as to wether or not you can reliably run multiple TDM400P's in the same machine? I need 4 x FXO and 4 x FXS to connect to both the PSTN and existing key system, but I have seen several threads suggesting that this is not a supported configuration i have two tdm400p's (2xFXO, 6xFXS) in one desktop machine used as asterisk server for a small office (so the pc hardware is nothing special). This configuration is running since two weeks without any problems! Thanks, Marc. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk + door opener
[EMAIL PROTECTED] wrote: Maybe do a transfer to a dedicated extension, which calls the script with the system() command to open the door? Or use the feature keys for a blind transfer. Seems like it could work. Btw, what kind of door phone opener do you have? I've been looking for something similar... I don't have one already. But there are some plain analog (a/b) door stations on the market that acts as a normal analog telephone. But I think they have the functionality to switch a door opener (with dtmf tones) already built in. I'll have to have a closer look to them in the near future. (something linke that (in german): http://www.telefon-ocker.de/cgi-bin/his-webshop.pl?f=NRc=H012518t=temarticv=OR) The only thing I have yet is the door opener itself which is switched by hand at the moment. Greg -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Thomas Artner Sent: Sunday, February 12, 2006 9:39 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] asterisk + door opener Hi! I am new to asterisk and I'd like to know wheter the following scenario is possible: Someone press the Button on the door station. The door station dials lets say the extension 333. I take the call on 333 and talk with the person on the door. Now I'd like to activate the door opener by pressing some numbers on the analog telefon. Asterisk should now recognize that I pressed something to open the door and should execute a script which opens the door. My question is, is it possible to execute a script while i am talking with the person on the door, without hanging up before? Can anyone give me some hints where to start looking in the docu?! I only need to know how to execute a script when I press - lets say the * Button while i am talking. Opening the door with a bash script is already working. Thx very much in advance! Tom ___ --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 ___ --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 ___ --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] asterisk + door opener
Hi! I am new to asterisk and I'd like to know wheter the following scenario is possible: Someone press the Button on the door station. The door station dials lets say the extension 333. I take the call on 333 and talk with the person on the door. Now I'd like to activate the door opener by pressing some numbers on the analog telefon. Asterisk should now recognize that I pressed something to open the door and should execute a script which opens the door. My question is, is it possible to execute a script while i am talking with the person on the door, without hanging up before? Can anyone give me some hints where to start looking in the docu?! I only need to know how to execute a script when I press - lets say the * Button while i am talking. Opening the door with a bash script is already working. Thx very much in advance! Tom ___ --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] attended call transfer
Questions for the community: is an integrated transfer feature valuable to you? Yes, merging blind and attended transfer would be valuable for me! If so, would you be willing to put out a bounty? Maybe. Depends on how much it would be. Tom ___ --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] attended call transfer
Hi! I got this answer from the digium support: You may wish to use the attended transfer by either using hold or flashhook instead of the # features.conf attended transfer option. From a phone connected via a Zap channel, you would need to hit flash. Then enter the extension which you wish to transfer to. You can either hangup once it begins ringing or wait until the remote end answers. Once the remote ends answers you may announce the caller then hangup. This method works better than the attended transfer option available in the features.conf file. You must have your dial plan configured properly to allow for transfers. Dial plan configuration also falls under our Express Technical Support Service. Regards, Chris Hozian That means, that an attended transfer is possible as it would be liked in this mailing-list-thread. I tried to make call transfers with the flash button, but it doesnt work. threewaycalling and transfer is set to yes in my zapata.conf. But when I hit the flash-button - nothing happens. All incoming calls triggers a Dial() on all extensions with the Dial-Parameter t - so a call transfer should be possible. (Are here further configurations necesseary in my dialplan?) What am I doing wrong? Tom ___ --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] attended call transfer
Hi! I am new with asterisk and I have my first problem with the attended call transfer feature. When a call comes in, i take the call and i would like to transfer it. So I press the * button (mapped for the attended transfer in features.conf) and the number for the receiving extension. The receiving extension rings and the call can be taken there. So far so good. Now to my problem: If I hook on the handset BEFORE the receiving extension take the call, the caller from outside will be disconnected and the receiving extension stops ringing. Shouldn't the receiving extension keep on ringing until the call is taken? Independent of hooking on the handset or not! (as it is with the blind transfer feature) The incoming line and all of the extensions are POTS, connected on a tdm400p card. I use asterisk 1.2.4 and zaptel 1.2.3 Hope someone could help me. Thx, Tom ___ --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
