Re: [asterisk-users] test the email-list OT
Ok, so I was fooled :P On 8/12/07, Stephen Bosch [EMAIL PROTECTED] wrote: C F wrote: OMG, someone thought that it's for real. Wow. I don't think so. Read the sentence carefully: On 8/11/07, Trevor Peirce [EMAIL PROTECTED] wrote: C F wrote: No you cant. This message is being dropped as well. Shame. Seriously though I posted a new thread right after I posted that He got the joke. -Stephen- ___ --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] asterisk and telewell isdn hfc problem
Hi, I have debian etch 4.0 machine (2.6.18) with two TW-ISDN PCI (Hfc) cards. I use bristuff-0.3.0-PRE-1y-e (asterisk-1.2.17,libpri-1.2.4,zaptel-1.2.16). I also have patched zaphfc with zaphfc_0.4.0-test1_florz-13.diff.gz (I load module: insmod /usr/src/bristuff-0.3.0-PRE-1y-e/zaphfc/zaphfc.ko modes=1 debug=1). So i want to test two cards and make loop between them. So one card would be NT, another TE. My configurations: /etc/zaptel.conf: loadzone=lt defaultzone=lt span=1,1,3,ccs,ami bchan=1-2 dchan=3 span=2,0,3,ccs,ami bchan=4-5 dchan=6 /etc/asterisk/zapata.conf [channels] switchtype = euroisdn ; p2mp NT mode signalling = bri_net_ptmp echocancel=yes echotraining = 100 echocancelwhenbridged=yes immediate=yes group = 1 context=from-internal channel = 1-2 signalling = bri_cpe_ptmp group = 2 context=from-internal channel = 4-5 So I get this from asterisk logs: Aug 12 03:08:23 DEBUG[2997] chan_sip.c: build_route: Contact hop: sip:[EMAIL PROTECTED] Aug 12 03:08:23 VERBOSE[3046] logger.c: -- Executing Dial(SIP/100-081c6668, ZAP/g1/52040004|60) in new stack Aug 12 03:08:23 VERBOSE[3046] logger.c: -- Requested transfer capability: 0x00 - SPEECH Aug 12 03:08:23 VERBOSE[3046] logger.c: -- Called g1/52040004 Aug 12 03:08:25 DEBUG[3046] chan_zap.c: Set option AUDIO MODE, value: ON(1) on Zap/1-1 Aug 12 03:08:25 DEBUG[3046] chan_zap.c: Hangup: channel: 1 index = 0, normal = 17, callwait = -1, thirdcall = -1 Aug 12 03:08:25 DEBUG[3046] chan_zap.c: Not yet hungup... Calling hangup once with icause, and clearing call Aug 12 03:08:25 DEBUG[3046] chan_zap.c: disabled echo cancellation on channel 1 Aug 12 03:08:25 DEBUG[3046] chan_zap.c: Set option TDD MODE, value: OFF(0) on Zap/1-1 Aug 12 03:08:25 DEBUG[3046] chan_zap.c: Updated conferencing on 1, with 0 conference users Aug 12 03:08:25 DEBUG[3046] chan_zap.c: Set option AUDIO MODE, value: OFF(0) on Zap/1-1 Aug 12 03:08:25 DEBUG[3046] chan_zap.c: disabled echo cancellation on channel 1 Aug 12 03:08:25 VERBOSE[3046] logger.c: -- Hungup 'Zap/1-1' Aug 12 03:08:25 DEBUG[3046] app_dial.c: Exiting with DIALSTATUS=CANCEL. And kernel log: Aug 12 11:01:25 pbx kernel: zaphfc: Card 0 configured for NT mode Aug 12 11:01:25 pbx kernel: zaphfc: Card 0 configured for master mode Aug 12 11:01:25 pbx kernel: ACPI: PCI Interrupt :04:01.0[A] - GSI 17 (level, low) - IRQ 177 Aug 12 11:01:25 pbx kernel: zaphfc: CCD/Billion/Asuscom 2BD0 configured at mem 0xe01faf00 fifo 0xdaef(0x1aef) IRQ 177 HZ 250 Aug 12 11:01:25 pbx kernel: zaphfc: Card 1 configured for TE mode Aug 12 11:01:25 pbx kernel: zaphfc: Card 1 configured for master mode Aug 12 11:01:25 pbx kernel: zaphfc: 2 hfc-pci card(s) in this box. Aug 12 11:01:25 pbx kernel: Registered tone zone 22 (Lithuania) Aug 12 03:08:25 pbx kernel: zaphfc[1]: b channel buffer underrun: 1, 0 Aug 12 03:08:25 pbx kernel: zaphfc[1]: b channel buffer overflow: 24, 24 Any ideas , how to fix this problem? Thanks ___ --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 telewell isdn hfc problem
On Sun, Aug 12, 2007 at 02:07:15AM +0300, Giedrius Augys wrote: Hi, I have debian etch 4.0 machine (2.6.18) with two TW-ISDN PCI (Hfc) cards. I use bristuff-0.3.0-PRE-1y-e (asterisk-1.2.17,libpri-1.2.4,zaptel-1.2.16). I also have patched zaphfc with zaphfc_0.4.0-test1_florz-13.diff.gz (I load module: insmod /usr/src/bristuff-0.3.0-PRE-1y-e/zaphfc/zaphfc.ko modes=1 debug=1). So i want to test two cards and make loop between them. So one card would be NT, another TE. My configurations: /etc/zaptel.conf: loadzone=lt defaultzone=lt span=1,1,3,ccs,ami bchan=1-2 dchan=3 span=2,0,3,ccs,ami bchan=4-5 dchan=6 /etc/asterisk/zapata.conf [channels] switchtype = euroisdn ; p2mp NT mode signalling = bri_net_ptmp echocancel=yes echotraining = 100 echocancelwhenbridged=yes immediate=yes group = 1 context=from-internal channel = 1-2 signalling = bri_cpe_ptmp group = 2 context=from-internal channel = 4-5 Are the spans actually up? What is the ouput of: pri show span 1 pri show span 2 -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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 telewell isdn hfc problem
2007/8/12, Tzafrir Cohen [EMAIL PROTECTED]: On Sun, Aug 12, 2007 at 02:07:15AM +0300, Giedrius Augys wrote: Hi, I have debian etch 4.0 machine (2.6.18) with two TW-ISDN PCI (Hfc) cards. I use bristuff-0.3.0-PRE-1y-e (asterisk-1.2.17,libpri-1.2.4,zaptel-1.2.16). I also have patched zaphfc with zaphfc_0.4.0-test1_florz-13.diff.gz (I load module: insmod /usr/src/bristuff-0.3.0-PRE-1y-e/zaphfc/zaphfc.ko modes=1 debug=1). So i want to test two cards and make loop between them. So one card would be NT, another TE. My configurations: /etc/zaptel.conf: loadzone=lt defaultzone=lt span=1,1,3,ccs,ami bchan=1-2 dchan=3 span=2,0,3,ccs,ami bchan=4-5 dchan=6 /etc/asterisk/zapata.conf [channels] switchtype = euroisdn ; p2mp NT mode signalling = bri_net_ptmp echocancel=yes echotraining = 100 echocancelwhenbridged=yes immediate=yes group = 1 context=from-internal channel = 1-2 signalling = bri_cpe_ptmp group = 2 context=from-internal channel = 4-5 Are the spans actually up? What is the ouput of: pri show span 1 pri show span 2 -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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 This is more information, I think it helps you: pbx*CLI pri show span 1 Primary D-channel: 3 Status: Provisioned, Up, Active Switchtype: EuroISDN Type: Network (PtMP) Overlap Dial: 0 T200 Timer: 1000 T203 Timer: 1 T305 Timer: 3 T308 Timer: 4000 T313 Timer: 4000 N200 Counter: 3 pbx*CLI pri show span 2 Primary D-channel: 6 Status: Provisioned, Down, Active Switchtype: EuroISDN Type: CPE (PtMP) Window Length: 0/7 Sentrej: 0 SolicitFbit: 0 Retrans: 0 Busy: 0 Overlap Dial: 0 T200 Timer: 1000 T203 Timer: 1 T305 Timer: 3 T308 Timer: 4000 T313 Timer: 4000 N200 Counter: 3 pbx:/etc/asterisk# cat /proc/zaptel/* Span 1: ZTHFC1 HFC-S PCI A ISDN card 0 [NT] layer 1 ACTIVATED (G3) AMI/CCS 1 ZTHFC1/0/1 Clear (In use) 2 ZTHFC1/0/2 Clear (In use) 3 ZTHFC1/0/3 HDLCFCS (In use) Span 2: ZTHFC2 HFC-S PCI A ISDN card 1 [TE] layer 1 DEACTIVATED (F5) AMI/CCS 4 ZTHFC2/0/1 Clear (In use) 5 ZTHFC2/0/2 Clear (In use) 6 ZTHFC2/0/3 HDLCFCS (In use) ___ --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] Dialplan loop
Am Donnerstag, den 09.08.2007, 20:12 -0500 schrieb David Bandel:
Folks,
I'm trying to implement a simple loop in a dialplan. The object is to
set a counter, run through some IVR options, increment the counter,
return to the start, then finally fall through to an operator or
voicemail.
exten = s,n,Set(loop = 0)
...
exten = s,n,Set(loop = $[${loop} + 1])
The above loop increment doesn't work. The error message is:
WARNING[14490]: ast_expr2.fl:398 ast_yyerror: ast_yyerror(): syntax
error: syntax error, unexpected '+', expecting $end; Input:
+ 1
^
WARNING[14490]: ast_expr2.fl:402 ast_yyerror: If you have questions,
please refer to doc/channelvariables.txt in the asterisk source.
Try removing extra space characters around the =. Very similar example
from my dialplan
exten = _2XX,n,Set(I=1)
...
exten = _2XX,n,Set(EXTR=$[${I} + 1])
Works fine. Also assigning a variable a new value based on the old value
works OK here (although not calculated, but concatenated):
exten = _2XX,n,Set(D=${D}SIP/sip501)
I am using Asterisk 1.2 here, but I remember similar errors with stray
characters.
BR
Anselm
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Re: [asterisk-users] Call forward at telco
Hello Gordon,
Thursday, August 9, 2007, 4:39:44 PM, you wrote:
This doesn't work?
exten = _*21*X.,1,Dial(Zap/1/*21*${EXTEN:4})
Then you can dial
*21*destination#
No that doesn't work. You can't dial this number. You have to send
special facility keypads to telco switch. Normal dialing would
signalling as called number, not as facility keypads.
pri debug span with called number (5 here):
4 Protocol Discriminator: Q.931 (8) len=8
4 Call Ref: len= 1 (reference 4/0x4) (Originator)
4 Message type: INFORMATION (123)
4 [70 02 81 35]
4 Called Number (len= 4) [ Ext: 1 TON: Unknown Number Type (0) NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1) '5' ]
4 -- Processing IE 112 (cs0, Called Party Number)
pri debug span with facility keypad (a * in this case):
4 Protocol Discriminator: Q.931 (8) len=7
4 Call Ref: len= 1 (reference 1/0x1) (Originator)
4 Message type: INFORMATION (123)
4 [2c 01 2a]
4 Keypad Facility (len= 1) [ *DÏÈN ]
4 -- Processing IE 44 (cs0, Keypad Facility)
ZapSendKeypadFacility in Asterisk 1.4 does this IN a call. But I do
not have a call. I have to pick up the line and send the information.
After # at the and a voice is telling me service activated or try
again.
Gunnar
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Re: [asterisk-users] asterisk and telewell isdn hfc problem
On Sun, Aug 12, 2007 at 12:10:50PM +0300, Giedrius Augys wrote: 2007/8/12, Tzafrir Cohen [EMAIL PROTECTED]: Are the spans actually up? What is the ouput of: pri show span 1 pri show span 2 This is more information, I think it helps you: pbx*CLI pri show span 1 Primary D-channel: 3 Status: Provisioned, Up, Active Switchtype: EuroISDN Type: Network (PtMP) Overlap Dial: 0 T200 Timer: 1000 T203 Timer: 1 T305 Timer: 3 T308 Timer: 4000 T313 Timer: 4000 N200 Counter: 3 pbx*CLI pri show span 2 Primary D-channel: 6 Status: Provisioned, Down, Active Switchtype: EuroISDN Type: CPE (PtMP) Window Length: 0/7 Sentrej: 0 SolicitFbit: 0 Retrans: 0 Busy: 0 Overlap Dial: 0 T200 Timer: 1000 T203 Timer: 1 T305 Timer: 3 T308 Timer: 4000 T313 Timer: 4000 N200 Counter: 3 pbx:/etc/asterisk# cat /proc/zaptel/* Span 1: ZTHFC1 HFC-S PCI A ISDN card 0 [NT] layer 1 ACTIVATED (G3) AMI/CCS 1 ZTHFC1/0/1 Clear (In use) 2 ZTHFC1/0/2 Clear (In use) 3 ZTHFC1/0/3 HDLCFCS (In use) Span 2: ZTHFC2 HFC-S PCI A ISDN card 1 [TE] layer 1 DEACTIVATED (F5) AMI/CCS 4 ZTHFC2/0/1 Clear (In use) 5 ZTHFC2/0/2 Clear (In use) 6 ZTHFC2/0/3 HDLCFCS (In use) Layer 1 activated on the NT port but deactivated on the NT port. Hmmm... what cable do you use between the two? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] LumenVox Speech Recognition
mitcheloc wrote: Nitesh, They claim to support numbers on their website so I would say yes. On 8/11/07, Nitesh Divecha [EMAIL PROTECTED] wrote: Dean, Can the LumenVox Speech Recognition engine understand numbers? Sorry to ask stupid questions but kinda curious... as for my application all I want is to the software to understand the numbers and provide me the output. Cheers, Nitesh Dean Collins wrote: No they have a standalone solution - lol NLVR is a whole separate server (or server farm) in most onsite installations. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial). -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of mitcheloc Sent: Saturday, 11 August 2007 7:29 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] LumenVox Speech Recognition Dean, Hmm.. I was hoping something that could be used with Asterisk on the machine locally... Nuance doesn't seem to offer that. On 8/11/07, Dean Collins [EMAIL PROTECTED] wrote: Nuance etc. and Steve to answer your questions - lumenvox just doesn't have the engine or phonetic capabilities that some of the the larger systems have. Like I said before - I've been stunned considering how cheap it is how good it is but. if you are looking for a less defined utterance structure it has limitations. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial). -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of mitcheloc Sent: Saturday, 11 August 2007 3:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] LumenVox Speech Recognition Dean, Are you aware of any better options for speech recognition? (though I'm sure more expensive) On 8/11/07, Nitesh Divecha [EMAIL PROTECTED] wrote: Thanks Dean... will update you on the progress... Cheers, Nitesh Dean Collins wrote: Hi Nitesh - yep great place to start. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial). -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Nitesh Divecha Sent: Saturday, 11 August 2007 11:40 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] LumenVox Speech Recognition Thanks Dean and Steve, I am planning to use for my IVR notification application which is built using PHPAGI and A2Billing (Callback, Calling Card). I saw the $50.00 Starter kit does it provide some functionality? Cheers, Nitesh Dean Collins wrote: Hi Steve, no I'm no expert at all I do however (or did) have an interest in building a far more comprehensive solution for an ASP solution combining other solutions that would have helped the asterisk community however could never get it off the ground. Nitesh to answer your original question...Lumenvox is great value for the money and works well - however there are limitations but for 90% of applications will work great. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial). -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Saturday, 11 August 2007 10:55 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] LumenVox Speech Recognition Dean Collins is probably the list expert on this. Thanks, Steve Totaro Nitesh Divecha wrote: Hello All, While looking for solution to solve my Callback DTMF problem, I came across LumenVox Speech Recognition software. Has anyone tried out? Need some feedback before I purchase We purchased a copy of the starter developer kit and got it going and yes it does recognized numbers. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com--
[asterisk-users] New Pico-ITX
Powerful enough to run a small asterisk server though not sure if the drop down in size from a mini-atx or a micro-atx but I'm sure someone will try. http://www.geek.com/first-look-via-px1-pico-itx-motherboard Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] +1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial). ___ --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] Playback a video file?
Is it possible to record or playback a video file in Asterisk? N. ___ --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] How to use OpenVPN with Asterisk
Kate Kretz wrote: OpenVPN is very good in NAT (if one of your boxes is behind NAT). otherwise, OpenVPN seems to be a bad choice, it's complicated, non-standard (there'n no RFC on OpenVPN). It's complicated? No more so then Asterisk. We use, it was quite straight forward to set up and the audio quality over it is quite good. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --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] Free sitting
Am Freitag, den 10.08.2007, 09:02 +0200 schrieb Olivier: Hi, My question is more what should be done than how should it be done. I could say : If you were a teacher, teaching and preparing your courses once a week (as you can't be called while teaching, can you ?) Well, yes. It always depends ;-) In an English or Arts course you could probably answer the phone to internal calls - those calling you will know you are in class and keep it as short as possible and just call instead of knocking on the door, which probably disturbs pretty much to the same amount. Getting external calls should then be turned off, or silent-ringer with a display showing external call and the send to voicemail button available. I assume that answering the phone while teaching the usage of circular saw and all those tools in a woodworks course or while teaching martial arts would be a bit too disturbing to make it happen ;-) would you prefer your phone system to log you in or out 1- automatically according a schedule stored somewhere, 2- whenever you turn your PC or or off, 3- when you dial something (for login) and logout) is done during nightimes, 4- when you dial something (for login and logout). 3/ and 4/ are compatible. You could further reason wether a user shall be logged out when the next one logs in. Logging the user out from a place when he logs in somewhere else is also reasonable (as you write below). Those two are even compatible with 2/ if only the login procedure shall login the phone, or only with 4/ if the logout is also coupled to the phone. My vote would go for the last one as it somehow keeps users responsible for themselves. A colleague prefers the third choice. Which would you pick ? If someone logs in from one place and logs in once again from somewhere else, then user previous log shall be replaced by the new one : incoming calls rings new phone. I'm wondering whether or not, 2 people could share the same phone but beside calling features, many supporting features such as MWI, BLF wouldn't it easily. Right. This depends on wether it will be a very seldom or a common case. Example a: There is a teachers' room where they usually sit in their non-teaching time and prepare lessons. Every place has (possibly a computer and a) phone. Example b: The same room has only one phone. Thinking about the computer coupling, that probably also depends on wether they regularly use the PC (all the time, part of the time, sometimes...) What do you think ? I would go for a combination of your 3/ and 4/ settings above. Allow them to logout, and if they do not, autologout after 3 hours or so (teachers probably not too often stay within the same room for more than three hours) or whenever they logout manually. You could combine that someone (you) is logged into this phone with a lamp on the phone (although you probably need a patch to asterisk to support non-regular presence/status settings) - perhaps making that lamp blink for 15 minutes before auto-logout, or depending on the number of states that the phone supports, signal message-waiting or one of about 1000 others things. You could also designate conference room phones such that multiple users can be logged in (without MWI and further features) while teacher's room phones and classroom phones could be strictly single-user and therefore offer extended features. Depending on the phone it can display both CALLERID(num) and CALLERID(name). You could tweak that to change CALLERID(name) to for Mr. Peters, for example, so that the display will tell both the caller number and the callee name. With 1000 more options of course. Users often lack the ability to know what they want and precisely be able to tell that. Asking them about their usage habits, with well formulated questions, might reveal which of the methods is best for your setting. I am not a teacher, but have lots of them in the family, so I know that between schools there are huge differences in work habits and so on. As an external consultant you will have to ask those who will (have to) use the system you design. A friend of mine says, Linux is all about choice. Same here for asterisk, and you are the one to choose. Best regards, Anselm ___ --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] New Pico-ITX
On Sun, 12 Aug 2007, Dean Collins wrote: Powerful enough to run a small asterisk server though not sure if the drop down in size from a mini-atx or a micro-atx but I'm sure someone will try. http://www.geek.com/first-look-via-px1-pico-itx-motherboard That's pretty impressive! I'm sure it will run linux+asterisk just fine with a 1GHz Via C7 processor as there are many platforms out there using that combination (or the Via C3) The issue that I've had (and I guess other small developers) is putting a motherboard in a pretty case so that it doesn't look like a PC when you turn it round... The other thing is that some people just don't believe it - I did a demo for someone with my usual box - pizza box style with a micro ATX board (VIA C3, 1GHz, fanless) and they couldn't believe it would handle the 35 extensions their current box handles - which takes up 6U of a rack. (My unit would have been in a 2U rack case for them, but they were still very skeptical!) Gordon ___ --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] New Pico-ITX
Just mount your solution along with patch panel, switch and UPS in one of these http://www.hoffmanonline.com/images/uploaded_images/87426/101806/87787768.gif On 8/12/07, Gordon Henderson [EMAIL PROTECTED] wrote: On Sun, 12 Aug 2007, Dean Collins wrote: Powerful enough to run a small asterisk server though not sure if the drop down in size from a mini-atx or a micro-atx but I'm sure someone will try. http://www.geek.com/first-look-via-px1-pico-itx-motherboard That's pretty impressive! I'm sure it will run linux+asterisk just fine with a 1GHz Via C7 processor as there are many platforms out there using that combination (or the Via C3) The issue that I've had (and I guess other small developers) is putting a motherboard in a pretty case so that it doesn't look like a PC when you turn it round... The other thing is that some people just don't believe it - I did a demo for someone with my usual box - pizza box style with a micro ATX board (VIA C3, 1GHz, fanless) and they couldn't believe it would handle the 35 extensions their current box handles - which takes up 6U of a rack. (My unit would have been in a 2U rack case for them, but they were still very skeptical!) Gordon ___ --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 -- /* Andrew Latham LATHAMA (lay-th-ham-eh) [EMAIL PROTECTED] [EMAIL PROTECTED] */ ___ --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] New Pico-ITX
Gordon Henderson wrote: On Sun, 12 Aug 2007, Dean Collins wrote: Powerful enough to run a small asterisk server though not sure if the drop down in size from a mini-atx or a micro-atx but I'm sure someone will try. http://www.geek.com/first-look-via-px1-pico-itx-motherboard That's pretty impressive! I'm sure it will run linux+asterisk just fine with a 1GHz Via C7 processor as there are many platforms out there using that combination (or the Via C3) The issue that I've had (and I guess other small developers) is putting a motherboard in a pretty case so that it doesn't look like a PC when you turn it round... The other thing is that some people just don't believe it - I did a demo for someone with my usual box - pizza box style with a micro ATX board (VIA C3, 1GHz, fanless) and they couldn't believe it would handle the 35 extensions their current box handles - which takes up 6U of a rack. (My unit would have been in a 2U rack case for them, but they were still very skeptical!) Gordon Just point them to the specs on a 3Com V3000. 1U and I think 200 extensions without having to add anything to it. Obviously, four FXO ports will not do much good for 200 extensions though. Thanks, Steve ___ --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] Shared Line Appearance - Aastra 55i - Does it work?
Does anyone have Shared (bridged) Line Appearance working in Asterisk 1.4?
Specifically with the Aastra 55i.
Specifically, I am using the Aastra 55i with the expansion module.
We want to see if other handsets are being used. (BLF) Getting BLF to work
would be a great start. It sounds like setting up the hints properly will
achieve this. right? Not totally sure how this should be configured.
We also want bridged appearances. Shared Line Appearance in Asterisk 1.4.
It is my understanding that with a bridged appearance, the line would show
as busy if it is in use on another handset, right? Meaning that the BLF
would be irrelevant?
in SLA.conf we have:
slatest]
type=trunk
device=SIP/1001
autocontext=slatest
[slatest1]
type=trunk
device=SIP/1003
autocontext=slatest1
[slateststation]
type=station
device=SIP/1002
autocontext=slateststation
trunk=slatest
trunk=slatest1
sip.conf
[1001]
type=friend
username=1001
secret=1001
host=dynamic
;context=slatest
context=slatest
dtmfmode=rfc2833
Language=en
qualify=yes
[EMAIL PROTECTED]
disallow=all
allow=all
[1002]
type=friend
username=1002
secret=1002
host=dynamic
;context=default1
context=slateststation
dtmfmode=rfc2833
Language=en
qualify=yes
[EMAIL PROTECTED]
disallow=all
allow=all
[1003]
type=friend
username=1003
secret=1003
host=dynamic
;context=default1
context=slatest1
dtmfmode=rfc2833
Language=en
qualify=yes
[EMAIL PROTECTED]
disallow=all
allow=all
Dialplan
[testing]
exten = _100X,1,Dial(SIP/${EXTEN}/${EXTEN})
exten = 101,1,Goto(slateststation|102|1)
exten = 102,1,Goto(slatest|1|1)
exten = 103,1,Goto(slatest1|1|1)
exten = h,1,Hangup()
[slatest]
exten = 1,1,SLATrunk(slatest)
exten = _X.,1,Dial(SIP/${EXTEN}/${EXTEN})
[slatest1]
exten = 1,1,SLATrunk(slatest1)
exten = _X.,1,Dial(SIP/${EXTEN}/${EXTEN})
[slateststation]
exten = 102,1,SLAStation(slateststation)
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Re: [asterisk-users] indications.c: Can't generate that much data!
Steve Totaro wrote: Dermot Bradley wrote: Linux 2.6.20, asterisk 1.2.23, mISDN 1_1_5, Digium B410P BRI card. When calls come in via ISDN the destination phone does ring but the caller hears no ringing tone, once the SIP phone is answered everything works as expected. Calls from SIP phone to SIP phone internally do let the caller hear a ringing tone. Do you have the r option in the line that dials the extension coming from the PSTN? As you know r option to Dial is not required to generate a ringback tone. If ringback tone is not being generated, adding r will seldom fix the issue. ___ --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] indications.c: Can't generate that much data!
Eric ManxPower Wieling wrote: Steve Totaro wrote: Dermot Bradley wrote: Linux 2.6.20, asterisk 1.2.23, mISDN 1_1_5, Digium B410P BRI card. When calls come in via ISDN the destination phone does ring but the caller hears no ringing tone, once the SIP phone is answered everything works as expected. Calls from SIP phone to SIP phone internally do let the caller hear a ringing tone. Do you have the r option in the line that dials the extension coming from the PSTN? As you know r option to Dial is not required to generate a ringback tone. If ringback tone is not being generated, adding r will seldom fix the issue. Maybe you do not understand my troubleshooting techniques. It was a question rather than a recommendation. After he responded that he did in fact have an r, I told him to try removing it. He did, problem solved, case closed. Thanks, Steve Totaro ___ --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] indications.c: Can't generate that much data!
Dermot Bradley wrote: I don't see what difference removing the r option has made from an Asterisk perspective - in both cases Asterisk tries to emulate a ringtone but fails for some reason when r is present. According to the the show application dial help having no r present for Dial should NOT generate a ringing tone yet here it does. You don't understand the docs. r will override whatever tone Asterisk thinks it should be providing and provide a ringing tone instead. Asterisk, by default, provides the correct tones to the caller. ___ --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] Ordering BRI From ATT
Trevor G. Hammonds wrote: I am not aware of any commercial Asterisk-compatible cards that support North American BRIs right out of the box. The best I have been able to come up with was a card sold on eBay, where the seller promises to supply a patch that needs to be applied to Asterisk (based on BRIstuff) so that it will support North American BRIs. The driver allows only one SPID per BRI, so multiple DID/MSNs are not supported. The card you're referring to is the OpenPCI card; they have a new stack that supports multiple SPIDs, which is now in beta testing. I understand that they actually have some cards deployed with US customers, too. Trevor -- are you using any BRIs at the moment? -Stephen- ___ --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] Ordering BRI From ATT
On Sun, Aug 12, 2007 at 12:42:10PM -0600, Stephen Bosch wrote: Trevor G. Hammonds wrote: I am not aware of any commercial Asterisk-compatible cards that support North American BRIs right out of the box. The best I have been able to come up with was a card sold on eBay, where the seller promises to supply a patch that needs to be applied to Asterisk (based on BRIstuff) so that it will support North American BRIs. The driver allows only one SPID per BRI, so multiple DID/MSNs are not supported. The card you're referring to is the OpenPCI card; Any relation between bristuff and chan_vpb that I wasn't aware of? they have a new stack that supports multiple SPIDs, which is now in beta testing. I understand that they actually have some cards deployed with US customers, too. Trevor -- are you using any BRIs at the moment? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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 Manager to Record Greetings
Am Freitag, den 10.08.2007, 11:26 -0500 schrieb Peder @ NetworkOblivion:
That's great, now say you have 5 or 6 AA's and each one has 10 different
parts that you want to record (thank you for calling... for Steve
press 1 for dave press 2). Rather than having to record a long
message, I want to break it into pieces so that if dave leaves, we can
just record that one chunk rather than the whole thing. I would need
lots of extensions pre-setup for each chunk. Not very efficient.
Gordon Henderson wrote:
On Fri, 10 Aug 2007, Peder @ NetworkOblivion wrote:
I am trying to use Asterisk Manager via php to record auto attendant
greetings and I just can't figure out how to do it. I've got the php
page working and I can click to call between two phones. However if I
click to call just a single phone and then try to enable monitor, when
I pick up the ringing phone, it just hangs up and doesn't record
anything. I'm sure I just don't know the appropriate syntax. Has
anybody done something like this? I can do the php stuff, I just need
the Asterisk Manager syntax.
I did something similar using multiple records in a row.
Something like
exten = 931,1,Answer()
exten = 931,2,Wait(2)
exten = 931,3,Set(E=1000)
exten = 931,4,Playback(beep)
exten = 931,5,Set(E=$[${E} + 1])
exten = 931,6,Record(/tmp/asterisk-recording-${E:1})
exten = 931,7,Playback(/tmp/asterisk-recording-${E:1})
exten = 931,8,Wait(2)
exten = 931,9,Goto(4)
This will loop: beep, record until # pressed, replay, wait, beep...
The files will be written with ascending numbers starting 001. Move
them to another place before doing the next recording session.
HTH
Anselm
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[asterisk-users] Converting an audio file to a .gsm format
Hello all, have anyone an idea about converting an audio file (.wav, .mp3, .au,...) to a .gsm audio file to use it as a voicemail file with Asterisk. Thanks. Abdelkader Mosbah ___ --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] indications.c: Can't generate that much data!
Eric ManxPower Wieling wrote: Dermot Bradley wrote: I don't see what difference removing the r option has made from an Asterisk perspective - in both cases Asterisk tries to emulate a ringtone but fails for some reason when r is present. According to the the show application dial help having no r present for Dial should NOT generate a ringing tone yet here it does. You don't understand the docs. r will override whatever tone Asterisk thinks it should be providing and provide a ringing tone instead. Asterisk, by default, provides the correct tones to the caller. I think he understood the docs perfectly. According to what you just said, there should have been ringing whether the r option was enabled or not (the only difference is whether ringing should be the correct indication). It is an obvious bug with a simple workaround that I helped him figure out. Thanks, Steve ___ --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 Manager to Record Greetings
Peder @ NetworkOblivion wrote: That's great, now say you have 5 or 6 AA's and each one has 10 different parts that you want to record (thank you for calling... for Steve press This is what I do. I found it some place on the wiki, it lets you record many prompts. exten = 4850,1,Goto(recordings,s,1) ; ** ; Welcome to the Audio prompt recording menu ; ** exten = s,1,Playback(local/extension-recording-menu) ; ; Please record your message, when ; completed press the # key ; exten = s,2,Playback(local/please-record-msg) exten = s,3,Record(mymessage:gsm) ; ; You said ; exten = s,4,Playback(local/you-said) exten = s,5,Playback(mymessage) ; *** ; Press 1 to continue or 2 to change your message ; *** exten = s,6,Background(local/press1-or-2) exten = s,7,Set(TIMEOUT(response)=2) exten = s,8,Set(TIMEOUT(digit)=2) exten = 1,1,System(/bin/mv /var/lib/asterisk/sounds/mymessage.gsm /var/lib/asterisk/sounds/local/`date +%s`.gsm) ; ; Thank you, your recording has been saved ; exten = 1,2,Playback(local/recording-saved) ; * ; Press 3 to record another message, or 4 to hangup ; * exten = 1,3,Background(local/press3-torecord-4tohang) exten = 2,1,Goto(recordings,s,2) exten = 3,1,Goto(recordings,s,2) exten = 4,1,Playback(vm-goodbye) exten = 4,2,Hangup() exten = t,1,Playback(local/sorry-didnot-getthat) exten = t,2,Goto(recordings,s,6) exten = i,1,Playback(local/sorry-invalid-choice) exten = i,2,Goto(recordings,s,2) Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --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] Converting an audio file to a .gsm format
MOSBAH ABDELKADER wrote: Hello all, have anyone an idea about converting an audio file (.wav, .mp3, .au,...) to a .gsm audio file to use it as a voicemail file with Asterisk. Thanks. Abdelkader Mosbah voip-info.org and google are your friends. http://www.voip-info.org/tiki-index.php?page=Convert+WAV+audio+files+for+use+in+Asterisk ___ --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] New Pico-ITX
I'm sure it will run linux+asterisk just fine with a 1GHz Via C7 processor as there are many platforms out there using that combination (or the Via C3) I recently gave up trying to use Jetway J7F2 motherboards (VIA C7 and VIA IDE/SATA/Ethernet) as they proved too unstable with Linux. I never managed to find the root cause - there's 1 box here that's rock solid and two others that keep locking up (only powercycle will clear) - one of the boxes locks up on average once per day. I'm not the only person having lockups on C7 systems either - there's chats on the VIA Linux forum about this. I'm now using Micro-ATX systems with Dual Core Athlons (the EE versions) - it ends up the same price to build as the Mini-ITX systems but has more horsepower and is better supported in Linux. VIA C3 C7 systems have been known to have DMA-related issues for some time. Stirk, Lamont Associates Ltd. Registered Address: Thomas Andrews House, Queens Road, Belfast, BT3 9DU Registered in Northern Ireland, Number: NI 47983. VAT Number: 832 2778 22 ___ --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] Converting an audio file to a .gsm format
On 8/12/07, MOSBAH ABDELKADER [EMAIL PROTECTED] wrote: have anyone an idea about converting an audio file (.wav, .mp3, .au,...) to a .gsm audio file to use it as a voicemail file with Asterisk. The program is called SoX. If you search voip-info.org you can find a nice guide with step-by-step instructions. I would suggest you not use the GSM format, unless you expect a majority of your calls to be in the GSM codec. ___ --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] 20min waiting time
I apologize if this question has already been answered / asked. I was searching on Google and nothing I do will work. All that happens is that the phones ring for 00:01:15 then voicemail kicks in. My goal here is to let the phones ring and ring until someone is not busy. I think 2 secs is long enough. Here is how the dial plan is setup exten=5,1,StartMusicOnHold exten=5,2,Dial(SIP/supportSIP/support2,2,tr) exten=5,3,VoiceMail([EMAIL PROTECTED]) exten=5,4,PlayBack(vm-goodbye) exten=5,5,HangUp() exten=1222,1,VoiceMailMain([EMAIL PROTECTED]) Any help is appreciated Otis ___ --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] indications.c: Can't generate that much data!
Luki wrote: Simple. When using r, asterisk needs to generate the ringing tones. For some reason your indicactions.conf describe a tone which is longer in duration than what can be generated by asterisk, so the error is shown and no tone is generated. Probably the max buffer length is somewhere preset in the code. I'm using stock indications.conf as shipped with Asterisk - if there was a fundamental mistake in that file then I'd expect the problem to occur with both mISDN and SIP channels, not just with mISDN. If you do NOT use the r flag, asterisk simply passes call progress indications from the source, without the need to generate any. Hence no error, and you hear ringing. But when r is present and when r is NOT present Asterisk still logs: DEBUG[13406] channel.c: Driver for channel 'mISDN/1-1' does not support indication 3, emulating it Which implies it generates the ringing tone in both situations, that's why I'm confused. Stirk, Lamont Associates Ltd. Registered Address: Thomas Andrews House, Queens Road, Belfast, BT3 9DU Registered in Northern Ireland, Number: NI 47983. VAT Number: 832 2778 22 ___ --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] Shared Line Appearance - Aastra 55i - Does it work?
Does anyone have Shared (bridged) Line Appearance working in Asterisk 1.4?
Specifically with the Aastra 55i.
Specifically, I am using the Aastra 55i with the expansion module.
We want to see if other handsets are being used. (BLF) Getting BLF to work
would be a great start. It sounds like setting up the hints properly will
achieve this. right? Not totally sure how this should be configured.
We also want bridged appearances. Shared Line Appearance in Asterisk 1.4.
It is my understanding that with a bridged appearance, the line would show
as busy if it is in use on another handset, right? Meaning that the BLF
would be irrelevant?
in SLA.conf we have:
slatest]
type=trunk
device=SIP/1001
autocontext=slatest
[slatest1]
type=trunk
device=SIP/1003
autocontext=slatest1
[slateststation]
type=station
device=SIP/1002
autocontext=slateststation
trunk=slatest
trunk=slatest1
sip.conf
[1001]
type=friend
username=1001
secret=1001
host=dynamic
;context=slatest
context=slatest
dtmfmode=rfc2833
Language=en
qualify=yes
[EMAIL PROTECTED]
disallow=all
allow=all
[1002]
type=friend
username=1002
secret=1002
host=dynamic
;context=default1
context=slateststation
dtmfmode=rfc2833
Language=en
qualify=yes
[EMAIL PROTECTED]
disallow=all
allow=all
[1003]
type=friend
username=1003
secret=1003
host=dynamic
;context=default1
context=slatest1
dtmfmode=rfc2833
Language=en
qualify=yes
[EMAIL PROTECTED]
disallow=all
allow=all
Dialplan
[testing]
exten = _100X,1,Dial(SIP/${EXTEN}/${EXTEN})
exten = 101,1,Goto(slateststation|102|1)
exten = 102,1,Goto(slatest|1|1)
exten = 103,1,Goto(slatest1|1|1)
exten = h,1,Hangup()
[slatest]
exten = 1,1,SLATrunk(slatest)
exten = _X.,1,Dial(SIP/${EXTEN}/${EXTEN})
[slatest1]
exten = 1,1,SLATrunk(slatest1)
exten = _X.,1,Dial(SIP/${EXTEN}/${EXTEN})
[slateststation]
exten = 102,1,SLAStation(slateststation)
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Re: [asterisk-users] Asterisk 1.2 TDM24xx and B410P
Hi here, Did you get any solution ? I've quiet the same pb : http://forums.digium.com/viewtopic.php?t=17394 Thank you for your answer. flo_turc ___ --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] Call file IAX Trunk: Call Failed, Reason 0
Hello, I am new to Asterisk; I did go through a lot of documentation, wikis, and the O'Reilly book and have most of what I need now working well. I do have a problem that I keep bumping heads against, however: I can dial out very well through a IAX2 trunk (9 followed by number), but if I specify the same IAX2 trunk in the Channel of a .call file, the call does not go through. Here is my test call file: Channel: IAX2/providername/14165551212 MaxRetries: 2 RetryTime: 20 WaitTime: 30 Application: Playback Data: hello-world The call is said to have failed with reason 0; the provider lists a 0-second call. Does anyone know what's going on ? (of course, the very same call file with an internal extension works perfectly well) Thank you ! ___ --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] TDM 2400 ?
Dear All, I'm using a TDM 2400 (12 FXO / 8 FXS) and I've some issues randomly : When receiving a call from PSTN and after Hang-up, Asterisk detect a second arrival call (same that channel) that is a ghost call off course. I don't PSTN suppervision with my telco... Any suggestions ? Flo_turc ___ --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] New Pico-ITX
On Sun, 12 Aug 2007, Dermot Bradley wrote: I'm sure it will run linux+asterisk just fine with a 1GHz Via C7 processor as there are many platforms out there using that combination (or the Via C3) I recently gave up trying to use Jetway J7F2 motherboards (VIA C7 and VIA IDE/SATA/Ethernet) as they proved too unstable with Linux. I never managed to find the root cause - there's 1 box here that's rock solid and two others that keep locking up (only powercycle will clear) - one of the boxes locks up on average once per day. I'm not the only person having lockups on C7 systems either - there's chats on the VIA Linux forum about this. I'm now using Micro-ATX systems with Dual Core Athlons (the EE versions) - it ends up the same price to build as the Mini-ITX systems but has more horsepower and is better supported in Linux. VIA C3 C7 systems have been known to have DMA-related issues for some time. Intersting/Worrying! I've not yet had any issues with the boards I've deployed - all CN1000's (Fanless 1GHz, Via C3 and my dev/test ones are older fanless 533MHz) ... Lots out in the field with Digium TDM400 or Sangoma analogue cards and Beronet ISDN2e cards (or none). # cat /proc/cpuinfo processor : 0 vendor_id : CentaurHauls cpu family : 6 model : 10 model name : VIA Esther processor 1000MHz stepping: 9 cpu MHz : 997.560 cache size : 128 KB fdiv_bug: no hlt_bug : no f00f_bug: no coma_bug: no fpu : yes fpu_exception : yes cpuid level : 1 wp : yes flags : fpu vme de pse tsc msr pae mce sep mtrr pge cmov pat clflush acpi mmx fxsr sse sse2 tm pni est tm2 rng rng_en ace ace_en ace2 ace2_en phe phe_en pmm pmm_en bogomips: 1996.56 # uptime 22:02:46 up 73 days, 10:59, 2 users, load average: 0.00, 0.00, 0.00 # asterisk -rx 'show uptime' System uptime: 8 weeks, 4 days, 12 hours, 18 minutes, 9 seconds Last reload: 7 weeks, 5 days, 8 hours, 53 minutes, 57 seconds And another: $ uptime 22:12:48 up 109 days, 9:19, 1 user, load average: 0.00, 0.00, 0.00 # rasterisk -rx 'show uptime' System uptime: 15 weeks, 4 days, 9 hours, 19 minutes, 52 seconds Last reload: 12 weeks, 4 days, 4 hours, 2 minutes, 23 seconds Verbosity is at least I've also built routers small NAS boxes out of the same motherboard... I compile up a custom kernel for them don't use the on-board audio hardware (nor printer port, but I use the serial ports on the routers!) and I always turn them off in the BIOS... So what am I doing wrong (or not as the case is!) Gordon ___ --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] 20min waiting time
OCOSA ListAcct wrote: I apologize if this question has already been answered / asked. I was searching on Google and nothing I do will work. All that happens is that the phones ring for 00:01:15 then voicemail kicks in. My goal here is to let the phones ring and ring until someone is not busy. I think 2 secs is long enough. Here is how the dial plan is setup exten=5,1,StartMusicOnHold exten=5,2,Dial(SIP/supportSIP/support2,2,tr) exten=5,3,VoiceMail([EMAIL PROTECTED]) exten=5,4,PlayBack(vm-goodbye) exten=5,5,HangUp() exten=1222,1,VoiceMailMain([EMAIL PROTECTED]) Any help is appreciated Otis Easiest way to solve your problem is to implement a support queue. Thanks, Steve ___ --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] Call file IAX Trunk: Call Failed, Reason 0
Please disregard. I have taken two debug dumps on a successful and a failed communication through the trunk; the only difference was in the absence of CallerID ! Now, this works well. I can say hello to the whole world. At 06:01 PM 8/12/2007, you wrote: Hello, I am new to Asterisk; I did go through a lot of documentation, wikis, and the O'Reilly book and have most of what I need now working well. I do have a problem that I keep bumping heads against, however: I can dial out very well through a IAX2 trunk (9 followed by number), but if I specify the same IAX2 trunk in the Channel of a .call file, the call does not go through. Here is my test call file: Channel: IAX2/providername/14165551212 MaxRetries: 2 RetryTime: 20 WaitTime: 30 Application: Playback Data: hello-world The call is said to have failed with reason 0; the provider lists a 0-second call. Does anyone know what's going on ? (of course, the very same call file with an internal extension works perfectly well) Thank you ! ___ --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
Re: [asterisk-users] LumenVox Speech Recognition
Nitesh, I've messed with the Lumenvox starter kit. If you are serious about this field, I think it's a must see. It was easy to set up and there are demos available. Their support is excellent. There is a quiet mailing list where questions are never ignored and most problems are solved AFAIK. Unfortunately, I have not had time to get to the next level of developing new demos for it, but I hope to do so some day. Take a look here for demos, etc. http://lumenvox.com/partners/integrator/digium/applicationzone/i /r ___ --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] FW: The trixbox Revolution Continues! Sign up for the Webinar.
On 8/8/07, Dean Collins [EMAIL PROTECTED] wrote: Hmm beginning of the end of free trixbox by the sounds of it. Dean, I thought you were on the conference call when Kerry discussed this in detail. There is no plan to dump the free version as I understood it. /r ps to all: the conference can be downloaded here: http://recordings.talkshoe.com/TC-22622/TS-38096.mp3 Trixbox news was one of the first items covered. ___ --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] LumenVox Speech Recognition
Randulo, There is an extra letter in the url you provided, it should be: http://lumenvox.com/partners/integrator/digium/applicationzone/ I think that the LumenVox pizza and weather demo would sound much better if the prompts were professionally recorded. On 8/12/07, randulo [EMAIL PROTECTED] wrote: Nitesh, I've messed with the Lumenvox starter kit. If you are serious about this field, I think it's a must see. It was easy to set up and there are demos available. Their support is excellent. There is a quiet mailing list where questions are never ignored and most problems are solved AFAIK. Unfortunately, I have not had time to get to the next level of developing new demos for it, but I hope to do so some day. Take a look here for demos, etc. http://lumenvox.com/partners/integrator/digium/applicationzone/i /r ___ --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 -- Mitchel Constantin Snap - A desktop user interface for Asterisk www.snapanumber.com ___ --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] 20min waiting time
Steve Totaro wrote: OCOSA ListAcct wrote: I apologize if this question has already been answered / asked. I was searching on Google and nothing I do will work. All that happens is that the phones ring for 00:01:15 then voicemail kicks in. My goal here is to let the phones ring and ring until someone is not busy. I think 2 secs is long enough. Here is how the dial plan is setup exten=5,1,StartMusicOnHold exten=5,2,Dial(SIP/supportSIP/support2,2,tr) exten=5,3,VoiceMail([EMAIL PROTECTED]) exten=5,4,PlayBack(vm-goodbye) exten=5,5,HangUp() exten=1222,1,VoiceMailMain([EMAIL PROTECTED]) Any help is appreciated Otis Easiest way to solve your problem is to implement a support queue. Queues in Asterisk are horrid little creatures. Many SIP phones and ITSPs will disconnect the call if the destination rings for a long time. Put an Answer as your first priority, this should fix your problem. ___ --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] 20min waiting time
Steve do you have an example that works for you. I am reading the queue literature nowThank you! Otis Steve Totaro wrote: OCOSA ListAcct wrote: I apologize if this question has already been answered / asked. I was searching on Google and nothing I do will work. All that happens is that the phones ring for 00:01:15 then voicemail kicks in. My goal here is to let the phones ring and ring until someone is not busy. I think 2 secs is long enough. Here is how the dial plan is setup exten=5,1,StartMusicOnHold exten=5,2,Dial(SIP/supportSIP/support2,2,tr) exten=5,3,VoiceMail([EMAIL PROTECTED]) exten=5,4,PlayBack(vm-goodbye) exten=5,5,HangUp() exten=1222,1,VoiceMailMain([EMAIL PROTECTED]) Any help is appreciated Otis Easiest way to solve your problem is to implement a support queue. Thanks, Steve ___ --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
Re: [asterisk-users] 20min waiting time
Eric so I should do this exten=5,1,Answer exten=5,2,StartMusicOnHold exten=5,3,Dial(SIP/supportSIP/support2,2,tr) exten=5,4,VoiceMail([EMAIL PROTECTED]) exten=5,5,PlayBack(vm-goodbye) exten=5,6,HangUp() exten=1222,1,VoiceMailMain([EMAIL PROTECTED]) Otis Eric ManxPower Wieling wrote: Steve Totaro wrote: OCOSA ListAcct wrote: I apologize if this question has already been answered / asked. I was searching on Google and nothing I do will work. All that happens is that the phones ring for 00:01:15 then voicemail kicks in. My goal here is to let the phones ring and ring until someone is not busy. I think 2 secs is long enough. Here is how the dial plan is setup exten=5,1,StartMusicOnHold exten=5,2,Dial(SIP/supportSIP/support2,2,tr) exten=5,3,VoiceMail([EMAIL PROTECTED]) exten=5,4,PlayBack(vm-goodbye) exten=5,5,HangUp() exten=1222,1,VoiceMailMain([EMAIL PROTECTED]) Any help is appreciated Otis Easiest way to solve your problem is to implement a support queue. Queues in Asterisk are horrid little creatures. Many SIP phones and ITSPs will disconnect the call if the destination rings for a long time. Put an Answer as your first priority, this should fix your problem. ___ --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] How strip +1 from caller id on inbound call
From some of our telecom providers we get the caller-id as:
NXXNXX
From others we get:
+1NXXNXX
We are trying to standardize the way our caller-id comes in so we
would like to strip off the +1 from the inbound caller id.
Can anyone offer any suggestions?
I have tried:
;exten = +18664918575,1,Set(CALLERID(all)=${CALLERIDNAME} ${CALLERIDNUM:2})
but it just yacks..
Thanks in advance for any help.
Regards,
Todd R.
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Live Answering Services
SIP or Toll-Free Connectivity
Light Accounts From $14.95/mo
http://www.PrestigeMessaging.com
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Re: [asterisk-users] How strip +1 from caller id on inbound call
voiplist wrote:
From some of our telecom providers we get the caller-id as:
NXXNXX
From others we get:
+1NXXNXX
We are trying to standardize the way our caller-id comes in so we
would like to strip off the +1 from the inbound caller id.
Can anyone offer any suggestions?
This is untested, but I think something like this ought to do it-
exten = s,n,ExecIf($[${CALLERID(num):0:1} = 1], Set,
CALLERID(num)=${CALLERID(num):1})
Trevor
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Re: [asterisk-users] How strip +1 from caller id on inbound call
you can do like this:
exten = _X.,1,GoSubIf($[${LEN(${CALLERID(num)})}10]?strip1);if it's
longer than grab the last 10 digits of the CIDNUM
exten =
_X.,50(strip1),Set(CALLERID(num)=${CALLERID(num):$[${LEN(${CALLERID(num)})}-10]});this
grabs the last 10 digits of CALLERID(num) and sets it to CALLERID(num)
exten = _X.,n,Return()
Hope this helps.
On 8/12/07, voiplist [EMAIL PROTECTED] wrote:
From some of our telecom providers we get the caller-id as:
NXXNXX
From others we get:
+1NXXNXX
We are trying to standardize the way our caller-id comes in so we
would like to strip off the +1 from the inbound caller id.
Can anyone offer any suggestions?
I have tried:
;exten = +18664918575,1,Set(CALLERID(all)=${CALLERIDNAME} ${CALLERIDNUM:2})
but it just yacks..
Thanks in advance for any help.
Regards,
Todd R.
--
Prestige Messaging
Live Answering Services
SIP or Toll-Free Connectivity
Light Accounts From $14.95/mo
http://www.PrestigeMessaging.com
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Re: [asterisk-users] How strip +1 from caller id on inbound call
After rereading this post, I belive that this could also be
acomplished doing this:
exten = _X.,1,GoSubIf($[${LEN(${CALLERID(num)})}10]?strip1);if it's
longer than 10 digits grab the last 10 digits of the CIDNUM
exten = _X.,50(strip1),Set(CALLERID(num)=${CALLERID(num):-10})
;this grabs the last 10 digits of CALLERID(num) and sets it to CALLERID(num)
exten = _X.,n,Return()
On 8/12/07, C F [EMAIL PROTECTED] wrote:
you can do like this:
exten = _X.,1,GoSubIf($[${LEN(${CALLERID(num)})}10]?strip1);if it's
longer than grab the last 10 digits of the CIDNUM
exten =
_X.,50(strip1),Set(CALLERID(num)=${CALLERID(num):$[${LEN(${CALLERID(num)})}-10]});this
grabs the last 10 digits of CALLERID(num) and sets it to CALLERID(num)
exten = _X.,n,Return()
Hope this helps.
On 8/12/07, voiplist [EMAIL PROTECTED] wrote:
From some of our telecom providers we get the caller-id as:
NXXNXX
From others we get:
+1NXXNXX
We are trying to standardize the way our caller-id comes in so we
would like to strip off the +1 from the inbound caller id.
Can anyone offer any suggestions?
I have tried:
;exten = +18664918575,1,Set(CALLERID(all)=${CALLERIDNAME}
${CALLERIDNUM:2})
but it just yacks..
Thanks in advance for any help.
Regards,
Todd R.
--
Prestige Messaging
Live Answering Services
SIP or Toll-Free Connectivity
Light Accounts From $14.95/mo
http://www.PrestigeMessaging.com
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Re: [asterisk-users] 20min waiting time
Eric ManxPower Wieling wrote: Steve Totaro wrote: OCOSA ListAcct wrote: I apologize if this question has already been answered / asked. I was searching on Google and nothing I do will work. All that happens is that the phones ring for 00:01:15 then voicemail kicks in. My goal here is to let the phones ring and ring until someone is not busy. I think 2 secs is long enough. Here is how the dial plan is setup exten=5,1,StartMusicOnHold exten=5,2,Dial(SIP/supportSIP/support2,2,tr) exten=5,3,VoiceMail([EMAIL PROTECTED]) exten=5,4,PlayBack(vm-goodbye) exten=5,5,HangUp() exten=1222,1,VoiceMailMain([EMAIL PROTECTED]) Any help is appreciated Otis Easiest way to solve your problem is to implement a support queue. Queues in Asterisk are horrid little creatures. Many SIP phones and ITSPs will disconnect the call if the destination rings for a long time. Put an Answer as your first priority, this should fix your problem. That is an odd statement about queues. I ran a call center handling over 15,000 calls a day using Asterisk and queues. No real problems. Please qualify your completely abstract statement, Queues in Asterisk are horrid little creatures. Statements like this are completely non productive to anyone. Thanks, Steve ___ --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] 20min waiting time
Steve Totaro wrote: Eric ManxPower Wieling wrote: Steve Totaro wrote: OCOSA ListAcct wrote: I apologize if this question has already been answered / asked. I was searching on Google and nothing I do will work. All that happens is that the phones ring for 00:01:15 then voicemail kicks in. My goal here is to let the phones ring and ring until someone is not busy. I think 2 secs is long enough. Here is how the dial plan is setup exten=5,1,StartMusicOnHold exten=5,2,Dial(SIP/supportSIP/support2,2,tr) exten=5,3,VoiceMail([EMAIL PROTECTED]) exten=5,4,PlayBack(vm-goodbye) exten=5,5,HangUp() exten=1222,1,VoiceMailMain([EMAIL PROTECTED]) Any help is appreciated Otis Easiest way to solve your problem is to implement a support queue. Queues in Asterisk are horrid little creatures. Many SIP phones and ITSPs will disconnect the call if the destination rings for a long time. Put an Answer as your first priority, this should fix your problem. That is an odd statement about queues. I ran a call center handling over 15,000 calls a day using Asterisk and queues. No real problems. Please qualify your completely abstract statement, Queues in Asterisk are horrid little creatures. Statements like this are completely non productive to anyone. Thanks, Steve Sorry to reply to my own post but for clarification, we had four queues. English sales, English support, Spanish sales, Spanish Support. At peek times, we would have 200-300 agents logged in and 600 or so callers. This was usually when our ads were running during Jerry Springer or Judge Judy. I think his two agent single queue would work just fine. Add Queuemetrics which is free (I believe) for five or less agents and then you can actually get some reporting on how your support role is handled. Thanks, Steve Totaro ___ --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] 20min waiting time
Steve do you have an example of this... Otis Steve Totaro wrote: Eric ManxPower Wieling wrote: Steve Totaro wrote: OCOSA ListAcct wrote: I apologize if this question has already been answered / asked. I was searching on Google and nothing I do will work. All that happens is that the phones ring for 00:01:15 then voicemail kicks in. My goal here is to let the phones ring and ring until someone is not busy. I think 2 secs is long enough. Here is how the dial plan is setup exten=5,1,StartMusicOnHold exten=5,2,Dial(SIP/supportSIP/support2,2,tr) exten=5,3,VoiceMail([EMAIL PROTECTED]) exten=5,4,PlayBack(vm-goodbye) exten=5,5,HangUp() exten=1222,1,VoiceMailMain([EMAIL PROTECTED]) Any help is appreciated Otis Easiest way to solve your problem is to implement a support queue. Queues in Asterisk are horrid little creatures. Many SIP phones and ITSPs will disconnect the call if the destination rings for a long time. Put an Answer as your first priority, this should fix your problem. That is an odd statement about queues. I ran a call center handling over 15,000 calls a day using Asterisk and queues. No real problems. Please qualify your completely abstract statement, Queues in Asterisk are horrid little creatures. Statements like this are completely non productive to anyone. Thanks, Steve ___ --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
Re: [asterisk-users] 20min waiting time
Yes, but I have to be up very early in the morning and it is getting late. The answer priority will work for you in the meantime. If you want to investigate using real queues, let me know and I will help you set it up. Most of the stuff is on the Wiki but I will give you exact settings that should work on your setup. If you plan on growing or ever want to collect data on queues, then this is the way to go. Thanks, Steve OCOSA ListAcct wrote: Steve do you have an example of this... Otis Steve Totaro wrote: Eric ManxPower Wieling wrote: Steve Totaro wrote: OCOSA ListAcct wrote: I apologize if this question has already been answered / asked. I was searching on Google and nothing I do will work. All that happens is that the phones ring for 00:01:15 then voicemail kicks in. My goal here is to let the phones ring and ring until someone is not busy. I think 2 secs is long enough. Here is how the dial plan is setup exten=5,1,StartMusicOnHold exten=5,2,Dial(SIP/supportSIP/support2,2,tr) exten=5,3,VoiceMail([EMAIL PROTECTED]) exten=5,4,PlayBack(vm-goodbye) exten=5,5,HangUp() exten=1222,1,VoiceMailMain([EMAIL PROTECTED]) Any help is appreciated Otis Easiest way to solve your problem is to implement a support queue. Queues in Asterisk are horrid little creatures. Many SIP phones and ITSPs will disconnect the call if the destination rings for a long time. Put an Answer as your first priority, this should fix your problem. That is an odd statement about queues. I ran a call center handling over 15,000 calls a day using Asterisk and queues. No real problems. Please qualify your completely abstract statement, Queues in Asterisk are horrid little creatures. Statements like this are completely non productive to anyone. Thanks, Steve ___ --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 ___ --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] New Pico-ITX
Gordon Henderson wrote: On Sun, 12 Aug 2007, Dermot Bradley wrote: I'm sure it will run linux+asterisk just fine with a 1GHz Via C7 processor as there are many platforms out there using that combination (or the Via C3) I recently gave up trying to use Jetway J7F2 motherboards (VIA C7 and VIA IDE/SATA/Ethernet) as they proved too unstable with Linux. I never managed to find the root cause - there's 1 box here that's rock solid and two others that keep locking up (only powercycle will clear) - one of the boxes locks up on average once per day. I'm not the only person having lockups on C7 systems either - there's chats on the VIA Linux forum about this. Just because someone is using an old kernel or doesn't know what they are doing doesn't mean the hardware is bad. I've had very good success with dozens of different VIA boards (from the original mini-itx board up to current C7 models, the Jetway boards included). I'm now using Micro-ATX systems with Dual Core Athlons (the EE versions) - it ends up the same price to build as the Mini-ITX systems but has more horsepower and is better supported in Linux. VIA C3 C7 systems have been known to have DMA-related issues for some time. I've seen those issues, but have never experienced them myself. Intersting/Worrying! I've not yet had any issues with the boards I've deployed - all CN1000's (Fanless 1GHz, Via C3 and my dev/test ones are older fanless 533MHz) ... Lots out in the field with Digium TDM400 or Sangoma analogue cards and Beronet ISDN2e cards (or none). snip # uptime 22:02:46 up 73 days, 10:59, 2 users, load average: 0.00, 0.00, 0.00 # asterisk -rx 'show uptime' System uptime: 8 weeks, 4 days, 12 hours, 18 minutes, 9 seconds Last reload: 7 weeks, 5 days, 8 hours, 53 minutes, 57 seconds And another: $ uptime 22:12:48 up 109 days, 9:19, 1 user, load average: 0.00, 0.00, 0.00 # rasterisk -rx 'show uptime' System uptime: 15 weeks, 4 days, 9 hours, 19 minutes, 52 seconds Last reload: 12 weeks, 4 days, 4 hours, 2 minutes, 23 seconds Verbosity is at least I've also built routers small NAS boxes out of the same motherboard... I compile up a custom kernel for them don't use the on-board audio hardware (nor printer port, but I use the serial ports on the routers!) and I always turn them off in the BIOS... So what am I doing wrong (or not as the case is!) Gordon, I don't think you're doing anything wrong. My experience with these little creatures is similar to yours. They perform well under load and just plain work. Darrick -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com ___ --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] 20min waiting time
Ok thanks. I will finish reading and see if I have any questions I will post and wait until you answer thank you! Otis Steve Totaro wrote: Yes, but I have to be up very early in the morning and it is getting late. The answer priority will work for you in the meantime. If you want to investigate using real queues, let me know and I will help you set it up. Most of the stuff is on the Wiki but I will give you exact settings that should work on your setup. If you plan on growing or ever want to collect data on queues, then this is the way to go. Thanks, Steve OCOSA ListAcct wrote: Steve do you have an example of this... Otis Steve Totaro wrote: Eric ManxPower Wieling wrote: Steve Totaro wrote: OCOSA ListAcct wrote: I apologize if this question has already been answered / asked. I was searching on Google and nothing I do will work. All that happens is that the phones ring for 00:01:15 then voicemail kicks in. My goal here is to let the phones ring and ring until someone is not busy. I think 2 secs is long enough. Here is how the dial plan is setup exten=5,1,StartMusicOnHold exten=5,2,Dial(SIP/supportSIP/support2,2,tr) exten=5,3,VoiceMail([EMAIL PROTECTED]) exten=5,4,PlayBack(vm-goodbye) exten=5,5,HangUp() exten=1222,1,VoiceMailMain([EMAIL PROTECTED]) Any help is appreciated Otis Easiest way to solve your problem is to implement a support queue. Queues in Asterisk are horrid little creatures. Many SIP phones and ITSPs will disconnect the call if the destination rings for a long time. Put an Answer as your first priority, this should fix your problem. That is an odd statement about queues. I ran a call center handling over 15,000 calls a day using Asterisk and queues. No real problems. Please qualify your completely abstract statement, Queues in Asterisk are horrid little creatures. Statements like this are completely non productive to anyone. Thanks, Steve ___ --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 ___ --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
Re: [asterisk-users] 20min waiting time
Yes. MOST of the time you should not use Answer, but in this specific case it may solve your issue. OCOSA ListAcct wrote: Eric so I should do this exten=5,1,Answer exten=5,2,StartMusicOnHold exten=5,3,Dial(SIP/supportSIP/support2,2,tr) exten=5,4,VoiceMail([EMAIL PROTECTED]) exten=5,5,PlayBack(vm-goodbye) exten=5,6,HangUp() exten=1222,1,VoiceMailMain([EMAIL PROTECTED]) Otis Eric ManxPower Wieling wrote: Steve Totaro wrote: OCOSA ListAcct wrote: I apologize if this question has already been answered / asked. I was searching on Google and nothing I do will work. All that happens is that the phones ring for 00:01:15 then voicemail kicks in. My goal here is to let the phones ring and ring until someone is not busy. I think 2 secs is long enough. Here is how the dial plan is setup exten=5,1,StartMusicOnHold exten=5,2,Dial(SIP/supportSIP/support2,2,tr) exten=5,3,VoiceMail([EMAIL PROTECTED]) exten=5,4,PlayBack(vm-goodbye) exten=5,5,HangUp() exten=1222,1,VoiceMailMain([EMAIL PROTECTED]) Any help is appreciated Otis Easiest way to solve your problem is to implement a support queue. Queues in Asterisk are horrid little creatures. Many SIP phones and ITSPs will disconnect the call if the destination rings for a long time. Put an Answer as your first priority, this should fix your problem. ___ --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 ___ --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] 20min waiting time
Steve Totaro wrote: Steve Totaro wrote: Eric ManxPower Wieling wrote: Steve Totaro wrote: OCOSA ListAcct wrote: I apologize if this question has already been answered / asked. I was searching on Google and nothing I do will work. All that happens is that the phones ring for 00:01:15 then voicemail kicks in. My goal here is to let the phones ring and ring until someone is not busy. I think 2 secs is long enough. Here is how the dial plan is setup exten=5,1,StartMusicOnHold exten=5,2,Dial(SIP/supportSIP/support2,2,tr) exten=5,3,VoiceMail([EMAIL PROTECTED]) exten=5,4,PlayBack(vm-goodbye) exten=5,5,HangUp() exten=1222,1,VoiceMailMain([EMAIL PROTECTED]) Any help is appreciated Otis Easiest way to solve your problem is to implement a support queue. Queues in Asterisk are horrid little creatures. Many SIP phones and ITSPs will disconnect the call if the destination rings for a long time. Put an Answer as your first priority, this should fix your problem. That is an odd statement about queues. I ran a call center handling over 15,000 calls a day using Asterisk and queues. No real problems. Please qualify your completely abstract statement, Queues in Asterisk are horrid little creatures. Statements like this are completely non productive to anyone. Thanks, Steve Sorry to reply to my own post but for clarification, we had four queues. English sales, English support, Spanish sales, Spanish Support. At peek times, we would have 200-300 agents logged in and 600 or so callers. This was usually when our ads were running during Jerry Springer or Judge Judy. I think his two agent single queue would work just fine. Add Queuemetrics which is free (I believe) for five or less agents and then you can actually get some reporting on how your support role is handled. In your situation it seems that queues work well for you. When you have dedicated agents answering calls full time queues work well. In non-call shops people forget to log out of the queue, are away from their desk often, and otherwise just screw up many of the assumptions that the Asterisk queue system makes. This is in addition to the learning curve. For a low number of calls and/or non-dedicated agents, a little bit of dialplan logic can do everything someone needs with something that is massively more flexible. ___ --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] Can't HANGUP call or channel on 1.4.9
I've isolated this problem the furthest that I can, and I'm now convinced this
is a bug in asterisk.
I have a context in extensions.conf like so:
[my_context]
exten = _X.,1,AGI(my_agi|${EXTEN}|${CHANNEL})
exten = _X.,2,GOTO(my_other_context|${EXTEN}|1)
exten = h,1,DeadAGI(my_agi_cleanup)
For the purposes of this scenario, my_agi simply will try to HANGUP the channel
to avoid the call going to priority 2 and instead go to my_agi_cleanup. Try as
I might, I cannot hang up the channel from within the agi!
I originate the channel thusly:
CLI originate SIP/50 extension [EMAIL PROTECTED]
I have agi debug turned on etc.
I pick up the call, my_agi is called with the correct parameters.
my_agi writes EXEC HANGUP and sleeps for 10 seconds. Asterisk responds 200
result=-1. The phone is still on the line.
my_agi writes EXEC HANGUP SIP/50-12345 (whatever channel it's given) and sleeps
for 10 seconds. Asterisk responds 200 result=-1. The phone is still on the line.
When the script exits, asterisk goes to the second priority which goes to
my_other_context. This shouldn't happen!
I try writing a GOTO h|1 in my_agi after my two HANGUP commands. Asterisk
responds 200 result=0. This does go to h|1, but the channel doesn't hang up
until my_agi_cleanup exits. This is a possible ugly work-around if no other
solution can be found.
I try putting a HANGUP() as the second priority and moving the GOTO to the
third. This does hangup the channel and goes to the 'h' priority, but defeats
the whole purpose of priorities.
If I do a soft hangup SIP/50-12345 in the console while my_agi sleeping
(copying the channel from the HANGUP command feedback from agi debug), the
correct thing happens. The channel is hung up and it avoids priority 2. Why
soft hangup works, HANGUP() as a priotity works, and HANGUP or HANGUP chan
within the AGI doesn't work is beyond me. Asterisk reports the correct
arguments are passed and so on.
I also tried dialing into the context via a GOTO from a DID extension. The same
problem manifests itself with the only change being the channel is a IAX2 trunk
instead of a SIP phone.
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Re: [asterisk-users] 20min waiting time
Steve / Eric When configuring the queue I tested works fine but one issue. My agent auto logs off after I am done with the call. I tried ignoring that option in agents.conf no luckAlso the below with the Answer line does not work either...still stays on and ring about 1:15 secs then goes to voicemail Otis Eric ManxPower Wieling wrote: Steve Totaro wrote: Steve Totaro wrote: Eric ManxPower Wieling wrote: Steve Totaro wrote: OCOSA ListAcct wrote: I apologize if this question has already been answered / asked. I was searching on Google and nothing I do will work. All that happens is that the phones ring for 00:01:15 then voicemail kicks in. My goal here is to let the phones ring and ring until someone is not busy. I think 2 secs is long enough. Here is how the dial plan is setup exten=5,1,StartMusicOnHold exten=5,2,Dial(SIP/supportSIP/support2,2,tr) exten=5,3,VoiceMail([EMAIL PROTECTED]) exten=5,4,PlayBack(vm-goodbye) exten=5,5,HangUp() exten=1222,1,VoiceMailMain([EMAIL PROTECTED]) Any help is appreciated Otis Easiest way to solve your problem is to implement a support queue. Queues in Asterisk are horrid little creatures. Many SIP phones and ITSPs will disconnect the call if the destination rings for a long time. Put an Answer as your first priority, this should fix your problem. That is an odd statement about queues. I ran a call center handling over 15,000 calls a day using Asterisk and queues. No real problems. Please qualify your completely abstract statement, Queues in Asterisk are horrid little creatures. Statements like this are completely non productive to anyone. Thanks, Steve Sorry to reply to my own post but for clarification, we had four queues. English sales, English support, Spanish sales, Spanish Support. At peek times, we would have 200-300 agents logged in and 600 or so callers. This was usually when our ads were running during Jerry Springer or Judge Judy. I think his two agent single queue would work just fine. Add Queuemetrics which is free (I believe) for five or less agents and then you can actually get some reporting on how your support role is handled. In your situation it seems that queues work well for you. When you have dedicated agents answering calls full time queues work well. In non-call shops people forget to log out of the queue, are away from their desk often, and otherwise just screw up many of the assumptions that the Asterisk queue system makes. This is in addition to the learning curve. For a low number of calls and/or non-dedicated agents, a little bit of dialplan logic can do everything someone needs with something that is massively more flexible. ___ --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
Re: [asterisk-users] Ordering BRI From ATT
Tzafrir Cohen wrote: On Sun, Aug 12, 2007 at 12:42:10PM -0600, Stephen Bosch wrote: Trevor G. Hammonds wrote: I am not aware of any commercial Asterisk-compatible cards that support North American BRIs right out of the box. The best I have been able to come up with was a card sold on eBay, where the seller promises to supply a patch that needs to be applied to Asterisk (based on BRIstuff) so that it will support North American BRIs. The driver allows only one SPID per BRI, so multiple DID/MSNs are not supported. The card you're referring to is the OpenPCI card; Any relation between bristuff and chan_vpb that I wasn't aware of? No -- sorry, my mistake. I got the name wrong. The card is actually from PhonicEQ; there's a description of the card at quadbri.phoniceq.com. I actually don't know much about the stack. I think it's a patched libpri, actually. It's sounds interesting, though I haven't seen it personally. -Stephen- ___ --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] 20min waiting time
Eric ManxPower Wieling wrote: Steve Totaro wrote: OCOSA ListAcct wrote: I apologize if this question has already been answered / asked. I was searching on Google and nothing I do will work. All that happens is that the phones ring for 00:01:15 then voicemail kicks in. My goal here is to let the phones ring and ring until someone is not busy. I think 2 secs is long enough. Here is how the dial plan is setup exten=5,1,StartMusicOnHold exten=5,2,Dial(SIP/supportSIP/support2,2,tr) exten=5,3,VoiceMail([EMAIL PROTECTED]) exten=5,4,PlayBack(vm-goodbye) exten=5,5,HangUp() exten=1222,1,VoiceMailMain([EMAIL PROTECTED]) Any help is appreciated Otis Easiest way to solve your problem is to implement a support queue. Queues in Asterisk are horrid little creatures. Many SIP phones and ITSPs will disconnect the call if the destination rings for a long time. Put an Answer as your first priority, this should fix your problem. Couldn't one change the default timeout so that Dial() will ring for 2 seconds? Or will that have all kinds of other undesirable side effects? -Stephen- ___ --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] 20min waiting time
OCOSA ListAcct wrote:
I apologize if this question has already been answered / asked. I was
searching on Google and nothing I do will work. All that happens is that
the phones ring for 00:01:15 then voicemail kicks in.
I wonder if this is your phone deciding it has been ringing for long
enough and rejecting the call.
Perhaps a NoOp(DialStatus is ${DIALSTATUS}) would shed light on this
possibility?
Trevor
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