[asterisk-users] [Dahdi 2.4.0] Flash() hangs up
Hello
I need Asterisk (1.4.39.2) to simulate a flash hook (ie. hitting the
R key on European handsets) so I can put a call on hold, dial a
second number, and set up a conference call.
By default, linux/include/dahdi/kernel.h sets the flashtime to 750ms,
which appears to be too long for European telcos, as they seem to
expect a line cut of about 100ms.
After editing the DAHDI_DEFAULT_FLASHTIME accordingly, I
recompiled/upgraded Dahdi, and ran the script, but Flash() still hangs
up the call:
===
[from_fxo]
exten = s,1,Wait(2)
exten = s,n,Set(GLOBAL(CID)=${CALLERID(num)})
exten = s,n,Hangup()
exten = h,1,system(/var/tmp/test10.lua ${CID})
[callback]
exten = start,1,NoOp(In callback, CID is ${CID})
;how to wait until remote phone picked up?
exten = start,n,Wait(5)
exten = start,n,Answer()
exten = start,n,Playback(please-wait)
;HERE Dahdi hangs up instead of getting a dialtone
exten = start,n,Flash()
exten = start,n,Dial(Dahdi/1/{GSM})
exten = start,n,Wait(5)
exten = start,n,Hangup()
;0 - Failed (not busy or congested)
;1 - Hung up
;3 - Ring timeout
;5 - Busy
;8 - Congestion
exten = failed,1,NoOp(Reason call failed is ${REASON})
===
Has someone successfully gotten Asterisk/Dahdi to use two-way calling?
Should I use another function than Flash()?
Thank you.
--
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Re: [asterisk-users] Asterisk/Skype
Can anyone make it more clear please
Regards
Khaled Chehab
NGN Eng.
Description: xplorium
Operations Office - Lebanon
Office : +961 1 868686 ext 115
Mobile: +961 3 045212
E-mail: kche...@xplorium.com
MSN ID :khalidche...@hotmail.com
Web Site: http://www.xplorium.com
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Friday, February 25, 2011 11:30 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Asterisk/Skype
AFAIK, the issue here is not Skype or Gtalk. The Asterisk client isn't
really designed to easily transport messages during the call or otherwise.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of William
Stillwell
Sent: Friday, February 25, 2011 3:14 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Asterisk/Skype
I am assuming that goes the same for Gtalk chat messages too?
Or has nobody played with that?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Terry Wilson
Sent: Friday, February 25, 2011 3:16 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk/Skype
On Feb 25, 2011, at 2:06 PM, Khaled W. Chehab wrote:
There is no debug appears,
Even I set core set verbose to 9
And skype set debug on
And in the extensions.conf I used
[Account]
exten = s,1,Set(message=${SKYPE_CHAT_RECEIVE(k_chehab,fakhourypbx,30)})
exten = s,n,NoOp(Received message: ${message})
The dialplan application is only for receiving chat messages during an
actual call. If you want to receive messages from outside of a call, you
will have to use the manager interface and look for SkyeChatMessage events.
image001.png--
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Re: [asterisk-users] Need to buy the Digium card, to confirm
The X1 card should seat in the X4 or X8 slots. Check out: http://computer.howstuffworks.com/pci-express1.htm HTH Cassius Smith On 2/26/11 4:33 PM, bilal ghayyad bilmar...@yahoo.com wrote: Hi All; My server and its slots written in it the following so I need to know which card to order it (I need a card supporting 2 E1s): PCIE_G2_X4 PCIE_G2_X8 Actually I do not know what is meaning by G2. OK I tried to buy directly from the below link but I found it is mentioned that it is x1 and not x4 or x8 so how can I get x4 or x8? The link: http://store.digium.com/productview.php?product_code=TE220B Description for the product: Two (2) span digital T1/E1/J1/PRI PCI-Express x1 card So please advise what do to? Regards Bilal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Using voice modem as poor man's FXO in Asterisk 1.8
Hi all, I've tried researching this, and so far, have struggled to find any contemporary information on the issue, so I do apologise if asking this irritates people who have answered this before. I have managed to set up Asterisk 1.8 on the web server here. I have two softphones (Ekiga) able to communicate with it. So far so good. I'm now curious to see if I can link it with the PSTN phone line here. The web server in question is an Intel Atom system with a Mini-ITX motherboard. Its one and only PCI slot is occupied by a PCI ethernet card. So FXO card is not an option even if it were within budget. My options therefore look to be an external FXO device of some description (Ethernet or USB), or to use a voice modem. I fear external FXOs are going to be even more expensive than internal FXO cards. Now, I have here an old Maestro JetStream 56k modem here that does amongst other things, voice comms, and I have used it in the past as a telephone by plugging a headset into the front of it (and it was full duplex too if I recall correctly). I have also used it as an answering machine, with the audio being transmitted digitally over the RS232 link. So that to me suggests it is possible to get audio in to and out of the modem, either via a sound card or using the serial port. The web server has a sound card too (hard not to buy a motherboard with one these days). Apart from the lack of any hardware signal processing, it seems all the components are there. The server isn't particularly heavily loaded, and thus I see no reason why the machine wouldn't theoretically be able to handle the DSP in software … I've seen lesser hardware do quite sophisticated DSP in real-time. Now, I've hunted high and low for where this is configured. Some mailing list threads point me to the nonexistant /etc/asterisk/modems.conf. One points me to /etc/asterisk/phone.conf, but nothing there jumps out at me as being an obvious means for configuring a modem — nor can I find where it's documented on the Asterisk wiki. Where abouts should I look for documentation on configuring these modules? Regards, -- Stuart Longland (aka Redhatter, VK4MSL) .'''. Gentoo Linux/MIPS Cobalt and Docs Developer '.'` : . . . . . . . . . . . . . . . . . . . . . . .'.' http://dev.gentoo.org/~redhatter :.' I haven't lost my mind... ...it's backed up on a tape somewhere. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using voice modem as poor man's FXO in Asterisk 1.8
On Mon, 28 Feb 2011, Stuart Longland wrote: Now, I have here an old Maestro JetStream 56k modem here that does An external modem is a non-starter. If you have infinite time and your time is worth US$0 and you're doing it just for the thrill of it -- maybe. Ward Mundy crew seem to think this kit (http://nerdvittles.com/?p=720) is pretty hot stuff. Sangoma makes a 2 FXO port USB thingy that looks interesting. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using voice modem as poor man's FXO in Asterisk 1.8
HI, My understanding is that the modem won't work. I believe asterisk does not support. I wonder why you do not have the built in ethernet in your motherboard. You can spare your PCI slot for a proper FXO card and use USB-to-ethernet For a PCI FXO card, the cheapest will be X100 but be aware of the quality and compatibility. Or a better choice will be TDM400 Other alternative: Get a USB-FXO from Sangoma, expensive Get a working SPA3000 as FXO --- cheapest I believe Get a OBi100, out of stock at the moment. I also want to try Hope this is of help to you CK On Mon, Feb 28, 2011 at 10:12 AM, Stuart Longland redhat...@gentoo.orgwrote: Hi all, I've tried researching this, and so far, have struggled to find any contemporary information on the issue, so I do apologise if asking this irritates people who have answered this before. I have managed to set up Asterisk 1.8 on the web server here. I have two softphones (Ekiga) able to communicate with it. So far so good. I'm now curious to see if I can link it with the PSTN phone line here. The web server in question is an Intel Atom system with a Mini-ITX motherboard. Its one and only PCI slot is occupied by a PCI ethernet card. So FXO card is not an option even if it were within budget. My options therefore look to be an external FXO device of some description (Ethernet or USB), or to use a voice modem. I fear external FXOs are going to be even more expensive than internal FXO cards. Now, I have here an old Maestro JetStream 56k modem here that does amongst other things, voice comms, and I have used it in the past as a telephone by plugging a headset into the front of it (and it was full duplex too if I recall correctly). I have also used it as an answering machine, with the audio being transmitted digitally over the RS232 link. So that to me suggests it is possible to get audio in to and out of the modem, either via a sound card or using the serial port. The web server has a sound card too (hard not to buy a motherboard with one these days). Apart from the lack of any hardware signal processing, it seems all the components are there. The server isn't particularly heavily loaded, and thus I see no reason why the machine wouldn't theoretically be able to handle the DSP in software … I've seen lesser hardware do quite sophisticated DSP in real-time. Now, I've hunted high and low for where this is configured. Some mailing list threads point me to the nonexistant /etc/asterisk/modems.conf. One points me to /etc/asterisk/phone.conf, but nothing there jumps out at me as being an obvious means for configuring a modem — nor can I find where it's documented on the Asterisk wiki. Where abouts should I look for documentation on configuring these modules? Regards, -- Stuart Longland (aka Redhatter, VK4MSL) .'''. Gentoo Linux/MIPS Cobalt and Docs Developer '.'` : . . . . . . . . . . . . . . . . . . . . . . .'.' http://dev.gentoo.org/~redhatter :.' I haven't lost my mind... ...it's backed up on a tape somewhere. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
