Re: [asterisk-users] Dial Chan_local Usage
Pointing out a typo.. either in the mail or in the actual dialplan:- exten = s,1,dial(local/2...@dialplan/n) [dailplan] -- Regards, Prince Singh W: http://www.drishti-soft.com B: http://blog.drishti-soft.com On Tue, Jun 30, 2009 at 8:27 PM, Benny Amorsen benny+use...@amorsen.dkbenny%2buse...@amorsen.dk wrote: Elliot Murdock murdo...@gmail.com writes: I needed to answer the local call for any sound to pass through: [default] exten = s,1,dial(local/2...@dialplan/n) [dailplan] exten = 220,1,answer() exten = 220,2,saydigits(123) exten = 220,3,dial(SIP/120||m) From my understanding, the answer command only answers the local call, but the final dial at priority 3 will remain unanswered. I guess you could put it that way, but notice that the original caller will start paying the moment you Answer(). Playing sounds before Answer() is called early media. It is unfortunately not universally supported -- possibly because it is so easily abused. Just imagine having two sets of phones, both transmitting early media. That would mean free calls. /Benny ___ -- 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] Extension status as XML for an Aastra 57i
2009/6/30 Carlos Chavez cur...@telecomabmex.com On Tue, 2009-06-30 at 16:17 -0400, Jeremy Winder wrote: I'm in the process of converting our current hybrid key system to Asterisk and Aastra 57i phones. One of the features that seems to be a show stopper for almost everyone in the office is the inability to see who is on the phone. Can someone point in the right direction to setup an XML app on the phone so they can press a soft-button and get a list of extensions and their statuses? I know I can use BLF and the line 2-4 buttons; but there are a lot more then 3 other people working here and I'm planning on using those of parking lots. Any help will be greatly appreciated as I'm an Asterisk noob learning as fast as I can. The 57i phone has 6 soft buttons which can show the status of at least 16 phones (if you do not want to use the rest of the soft buttons which would give you another 16). Are you sure of that ? How can you set more than one single phone to light on or off a given BLF ? With a single button, I agree you can query more than one phone status but the associated light can't display more than one phone or am I missing something ? If you really need to have more you should use the 536M or 560M console which can display up to 60 extensions. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 ___ -- 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] Multi-tenant parking broken in 1.6.1.1?
Hello, all. With the assistance of very helpful folks, our brand new
multi-tenant setup seems to be working smoothly from start to finish
with just a bump or two. The biggest is parking. Now that we got most
kinks worked out, I'm a little more comfortable in trying to resolve
this.
There seem to be two problems:
1. Parking assigns parking spaces from the default group no matter
what we do.
2. When the parked call timer expires, the callback to the original
callee fails because a | delimiter is used in the Dial()
function.
Perhaps we have configured it incorrectly. Here is the pertinent
section from features.conf:
[parkinglot_a10] ; EBC
context = a10parking
parkpos = 101-110
;parkext = 100
findslot = next
[parkinglot_a100] ; SSI
context = a100parking
;parkext = 1000
parkpos = 1001-1020
findslot = next
If I understand this correctly, the parkinglog_a100 would be the channel
variable and a100parking the context into which parking extensions are
placed.
We set the channel parameter in sip.conf:
[a100](!,common)
context=a100
vmext=999
parkinglot=parkinglot_a100
subscribecontext=a100
accountcode=a-0100
fromdomain=ssiservices.biz
[userx](a100)
mailbox=...@a100,x...@a100
secret=something
callerid=John A. Sullivan III xxx
fromuser=userid
and we included the context in extensions.conf:
[a100] ; SSI
exten = 911,1,Macro(emergency-US,xx)
exten = 9911,1,Macro(emergency-US,xx)
exten = ,1,VoiceMailMain(${CALLERID(num)}...@a100) ; Direct mail
retrieval
include = a100pub
include = a100conf
include = a100parking
include = US-international
include = dial-uri
We also tried Set(CHANNEL(parkinglot)=parkinglot_a100). We also tried
creating our own parking which yielded interesting data but not
solution.
Here is the console output using the regular setup described:
Call comes in and is answered:
-- SIP/gss-cc01c918 answered SIP/localhost-cc002cf8
-- Native bridging SIP/localhost-cc002cf8 and SIP/gss-cc01c918
-- Started music on hold, class 'default', on SIP/localhost-cc002cf8
== Using SIP RTP TOS bits 176
== Using SIP RTP CoS mark 5
Call is parked:
-- Executing [...@a100:1] Park(SIP/gss-cc05ceb8, ) in new stack
== Parked SIP/gss-cc05ceb8 on 701 (lot default). Will timeout back to
extension [a100] s, 1 in 60 seconds
-- Added extension '701' priority 1 to parkedcalls (0x2cca3f70)
-- SIP/gss-cc05ceb8 Playing 'digits/7.ulaw' (language 'en')
-- SIP/gss-cc05ceb8 Playing 'digits/0.ulaw' (language 'en')
-- SIP/gss-cc05ceb8 Playing 'digits/1.ulaw' (language 'en')
-- Started music on hold, class 'default', on SIP/gss-cc05ceb8
I'm not sure what is happening here but I think this is the original
callee releasing the call. I don't know what the ZOMBIE extension is
about:
== Spawn extension (a100, s, 1) exited non-zero on
'Parked/SIP/gss-cc05ceb8ZOMBIE'
-- Auto fallthrough, channel 'Parked/SIP/gss-cc05ceb8ZOMBIE' status is
'UNKNOWN'
-- Executing [...@a100:1] Answer(Parked/SIP/gss-cc05ceb8ZOMBIE, 0.5)
in new stack
== Spawn extension (a100, h, 1) exited non-zero on
'Parked/SIP/gss-cc05ceb8ZOMBIE'
-- Stopped music on hold on SIP/gss-cc05ceb8
-- Stopped music on hold on SIP/localhost-cc002cf8
-- Started music on hold, class 'default', on SIP/localhost-cc002cf8
== Spawn extension (macro-common, s, 1) exited non-zero on
'SIP/gss-cc05ceb8ZOMBIE' in macro 'common'
== Spawn extension (a100pub, 314, 2) exited non-zero on
'SIP/gss-cc05ceb8ZOMBIE'
== Using SIP RTP TOS bits 176
== Using SIP RTP CoS mark 5
Then we see the destination callee attempting to pick up the call and is
the output of our routine to catch misdialed/unknown extensions:
-- Executing [...@a100:1] GotoIf(SIP/jasiii-cc05ceb8, 0?:_.,1) in new
stack
-- Goto (a100,_.,1)
-- Executing [...@a100:1] Answer(SIP/jasiii-cc05ceb8, 0.5) in new stack
-- Executing [...@a100:2] Playback(SIP/jasiii-cc05ceb8, im-sorry) in
new stack
-- SIP/jasiii-cc05ceb8 Playing 'im-sorry.ulaw' (language 'en')
-- Executing [...@a100:3] Wait(SIP/jasiii-cc05ceb8, 0.0.5) in new stack
-- Executing [...@a100:4] Playback(SIP/jasiii-cc05ceb8,
you-dialed-wrong-number) in new stack
-- SIP/jasiii-cc05ceb8 Playing 'you-dialed-wrong-number.ulaw' (language
'en')
-- Executing [...@a100:5] Wait(SIP/jasiii-cc05ceb8, 0.4) in new stack
-- Executing [...@a100:6] Playback(SIP/jasiii-cc05ceb8, vm-goodbye) in
new stack
-- SIP/jasiii-cc05ceb8 Playing 'vm-goodbye.ulaw' (language 'en')
-- Executing [...@a100:7] Hangup(SIP/jasiii-cc05ceb8, ) in new stack
== Spawn extension (a100, _., 7) exited non-zero on 'SIP/jasiii-cc05ceb8'
-- Executing [...@a100:1] Answer(SIP/jasiii-cc05ceb8, 0.5) in new stack
== Spawn extension (a100, h, 1) exited non-zero on 'SIP/jasiii-cc05ceb8'
We then see the park timeout and fail to return to the original
Re: [asterisk-users] Puzzling problem
2 Things:- 1. Keep relevant subject line of the mails to public forums :) 2. Try direct IP call to another grandstream. On Wed, Jul 1, 2009 at 5:56 AM, Todd Reese trees...@gmail.com wrote: I did the upgrade to the phone. And the problem continued. Currently, as per the previous poster, I have reset the phone to the factory default and have started setup again. Peder wrote: Try upgrading the firmware on it. They have all sorts of goofy bugs. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Todd Reese Sent: Tuesday, June 30, 2009 4:56 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Puzzling problem Hi All, I have a problem with my Asterisk Server that the logs aren't giving me any clue to what's going on. The server is running 1.6.1.1 and connected to a Grandstream GXP2000 phone. At 3:58 minutes the call cuts off with no indication in the log. This is random and is only localized to that 1 phone. The other phone is a cordless connected through a Sipura Box with no problems. I've tried other versions of Asterisk after the problem started and it is continuing. Any help on where to look for clues is greatly appreciated. TIA, Todd Reese ___ -- 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 -- Regards, Prince Singh Drishti-Soft Solutions Pvt Ltd 62-A, First Floor, Maruti Industrial Area, Sector - 18, Gurgaon - 122016 Haryana, India. P: 91 124 4771000 F: 91 124 4039120 W: http://www.drishti-soft.com B: http://blog.drishti-soft.com DISCLAIMER This message may contain confidential, proprietary or legally Privileged information. In case you are not the original intended Recipient of the message, you must not, directly or indirectly, use, disclose, distribute, print, or copy any part of this message and you are requested to delete it and inform the sender. Any views expressed in this message are those of the individual sender unless otherwise stated. Nothing contained in this message shall be construed as an offer or acceptance of any offer by Drishti-Soft Solutions Pvt Ltd (Drishti) unless sent with that express intent and with due authority of Drishti. Drishti has taken enough precautions to prevent the spread of viruses. However the company accepts no liability for any damage caused by any virus transmitted by this email. ___ -- 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] Echo and static on PRI with errors
John F. Ervin wrote: What do you do if you find things sharing interrupts (IRQ 11) in my case with my X100P card. I believe there is some sort of internal audio card in my cheap slow PC. Check the BIOS whether you can: Change the IRQ assignments Disable the extra hardware using the same IRQ Or otherwise try changing the slot it is in... I had very good results in the past swapping card around Good luck! --FP ___ -- 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] Multi-tenant parking broken in 1.6.1.1?
On Wed, 2009-07-01 at 02:17 -0400, John A. Sullivan III wrote:
Hello, all. With the assistance of very helpful folks, our brand new
multi-tenant setup seems to be working smoothly from start to finish
with just a bump or two. The biggest is parking. Now that we got most
kinks worked out, I'm a little more comfortable in trying to resolve
this.
There seem to be two problems:
1. Parking assigns parking spaces from the default group no matter
what we do.
2. When the parked call timer expires, the callback to the original
callee fails because a | delimiter is used in the Dial()
function.
Perhaps we have configured it incorrectly. Here is the pertinent
section from features.conf:
[parkinglot_a10] ; EBC
context = a10parking
parkpos = 101-110
;parkext = 100
findslot = next
[parkinglot_a100] ; SSI
context = a100parking
;parkext = 1000
parkpos = 1001-1020
findslot = next
If I understand this correctly, the parkinglog_a100 would be the channel
variable and a100parking the context into which parking extensions are
placed.
We set the channel parameter in sip.conf:
[a100](!,common)
context=a100
vmext=999
parkinglot=parkinglot_a100
subscribecontext=a100
accountcode=a-0100
fromdomain=ssiservices.biz
[userx](a100)
mailbox=...@a100,x...@a100
secret=something
callerid=John A. Sullivan III xxx
fromuser=userid
and we included the context in extensions.conf:
[a100] ; SSI
exten = 911,1,Macro(emergency-US,xx)
exten = 9911,1,Macro(emergency-US,xx)
exten = ,1,VoiceMailMain(${CALLERID(num)}...@a100) ; Direct mail
retrieval
include = a100pub
include = a100conf
include = a100parking
include = US-international
include = dial-uri
We also tried Set(CHANNEL(parkinglot)=parkinglot_a100). We also tried
creating our own parking which yielded interesting data but not
solution.
Here is the console output using the regular setup described:
Call comes in and is answered:
-- SIP/gss-cc01c918 answered SIP/localhost-cc002cf8
-- Native bridging SIP/localhost-cc002cf8 and SIP/gss-cc01c918
-- Started music on hold, class 'default', on SIP/localhost-cc002cf8
== Using SIP RTP TOS bits 176
== Using SIP RTP CoS mark 5
Call is parked:
-- Executing [...@a100:1] Park(SIP/gss-cc05ceb8, ) in new stack
== Parked SIP/gss-cc05ceb8 on 701 (lot default). Will timeout back to
extension [a100] s, 1 in 60 seconds
-- Added extension '701' priority 1 to parkedcalls (0x2cca3f70)
-- SIP/gss-cc05ceb8 Playing 'digits/7.ulaw' (language 'en')
-- SIP/gss-cc05ceb8 Playing 'digits/0.ulaw' (language 'en')
-- SIP/gss-cc05ceb8 Playing 'digits/1.ulaw' (language 'en')
-- Started music on hold, class 'default', on SIP/gss-cc05ceb8
I'm not sure what is happening here but I think this is the original
callee releasing the call. I don't know what the ZOMBIE extension is
about:
== Spawn extension (a100, s, 1) exited non-zero on
'Parked/SIP/gss-cc05ceb8ZOMBIE'
-- Auto fallthrough, channel 'Parked/SIP/gss-cc05ceb8ZOMBIE' status is
'UNKNOWN'
-- Executing [...@a100:1] Answer(Parked/SIP/gss-cc05ceb8ZOMBIE,
0.5) in new stack
== Spawn extension (a100, h, 1) exited non-zero on
'Parked/SIP/gss-cc05ceb8ZOMBIE'
-- Stopped music on hold on SIP/gss-cc05ceb8
-- Stopped music on hold on SIP/localhost-cc002cf8
-- Started music on hold, class 'default', on SIP/localhost-cc002cf8
== Spawn extension (macro-common, s, 1) exited non-zero on
'SIP/gss-cc05ceb8ZOMBIE' in macro 'common'
== Spawn extension (a100pub, 314, 2) exited non-zero on
'SIP/gss-cc05ceb8ZOMBIE'
== Using SIP RTP TOS bits 176
== Using SIP RTP CoS mark 5
Then we see the destination callee attempting to pick up the call and is
the output of our routine to catch misdialed/unknown extensions:
-- Executing [...@a100:1] GotoIf(SIP/jasiii-cc05ceb8, 0?:_.,1) in new
stack
-- Goto (a100,_.,1)
-- Executing [...@a100:1] Answer(SIP/jasiii-cc05ceb8, 0.5) in new
stack
-- Executing [...@a100:2] Playback(SIP/jasiii-cc05ceb8, im-sorry) in
new stack
-- SIP/jasiii-cc05ceb8 Playing 'im-sorry.ulaw' (language 'en')
-- Executing [...@a100:3] Wait(SIP/jasiii-cc05ceb8, 0.0.5) in new
stack
-- Executing [...@a100:4] Playback(SIP/jasiii-cc05ceb8,
you-dialed-wrong-number) in new stack
-- SIP/jasiii-cc05ceb8 Playing 'you-dialed-wrong-number.ulaw' (language
'en')
-- Executing [...@a100:5] Wait(SIP/jasiii-cc05ceb8, 0.4) in new stack
-- Executing [...@a100:6] Playback(SIP/jasiii-cc05ceb8, vm-goodbye)
in new stack
-- SIP/jasiii-cc05ceb8 Playing 'vm-goodbye.ulaw' (language 'en')
-- Executing [...@a100:7] Hangup(SIP/jasiii-cc05ceb8, ) in new stack
== Spawn extension (a100, _., 7) exited non-zero on 'SIP/jasiii-cc05ceb8'
-- Executing
Re: [asterisk-users] Intercepting a Call while ringing a device
Here's how to configure this method properly http://www.voip-info.org/wiki/view/Asterisk+callgroups+and+pickupgroups Ish Danny Nicholas wrote: If it is configured and working correctly, *8 picks up the ringing line from any eligible phone. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Elliot Murdock Sent: Tuesday, June 30, 2009 2:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Intercepting a Call while ringing a device Hello! I am looking for a way to dynamically redirect a call while it is ringing to another device. Basically, if a person is far away from his desk, he should have the option to use another phone and pick up the call. Thanks for any suggestions, Elliot ___ -- 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 -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 ___ -- 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] Fwd: Unknown udp ports listening experts calling !
-- Forwarded message -- From: Xavier Cardil cardil.xav...@gmail.com Date: Wed, Jul 1, 2009 at 10:51 AM Subject: Unknown udp ports listening experts calling ! To: asterisk-users-requ...@lists.digium.com Hello, last days we run under an very heavy issue with one audio stream getting mixed with our RTP traffic. The audio source was unknown and changing the asterisks to other net interfaces and hooking them to another Vlan did the trick. The audio stream is not coming anymore so it is some outside UDP source sending data to that interface. On the way, we changed the asterisk UDP port range to 3-4 instead of the default 1-2. Can somebody tell me why asterisk still listening or transfering data through these ports ? I'm trying to solve the problem, as I find it very interesting. udp0 0 0.0.0.0:27270.0.0.0:* 4989/asterisk udp0 0 0.0.0.0:90010.0.0.0:* 26354/udp-sender udp0 0 0.0.0.0:50000.0.0.0:* 4989/asterisk 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
Re: [asterisk-users] Echo and static on PRI with errors
On Wed, Jul 1, 2009 at 7:37 AM, Francesco Peeters france...@fampeeters.comwrote: John F. Ervin wrote: What do you do if you find things sharing interrupts (IRQ 11) in my case with my X100P card. I believe there is some sort of internal audio card in my cheap slow PC. Check the BIOS whether you can: Change the IRQ assignments Disable the extra hardware using the same IRQ Or otherwise try changing the slot it is in... I had very good results in the past swapping card around Good luck! I did a bit of investigation WRT the IRQ settings on this box. 00:02.1 USB Controller: nVidia Corporation CK804 USB Controller (rev a3) (prog-if 20) Subsystem: Hewlett-Packard Company Device 3207 Flags: bus master, 66MHz, fast devsel, latency 0, IRQ 11 -- 01:05.0 VGA compatible controller: nVidia Corporation NV11 [GeForce2 MX/MX 400] (rev b2) Subsystem: Hewlett-Packard Company Device 3207 Flags: bus master, 66MHz, medium devsel, latency 64, IRQ 11 -- 02:00.0 Ethernet controller: Broadcom Corporation NetXtreme BCM5721 Gigabit Ethernet PCI Express (rev 11) Subsystem: Hewlett-Packard Company Device 3209 Flags: bus master, fast devsel, latency 0, IRQ 11 -- 81:01.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface Subsystem: Device 79fe:0001 Flags: bus master, medium devsel, latency 64, IRQ 11 So basically there's 2 network cards and a USB controller sharing IRQ 11 with the Openvox card. I wasn't able to find any settings in the bios to manually configure IRQ assignments :( Could someone tell me how to set which IRQ the ISDN card picks up? -- Tom O'Connor http://www.twinhelix.org t...@twinhelix.org ___ -- 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] Fwd: Unknown udp ports listening experts calling !
On 1 Jul 2009, at 09:54, Xavier Cardil wrote: udp0 0 0.0.0.0:2727 0.0.0.0:* 4989/asterisk udp0 0 0.0.0.0:9001 0.0.0.0:* 26354/udp-sender udp0 0 0.0.0.0:5000 0.0.0.0:* 4989/asterisk 2727 = mgcp I found that with Google. A useful tool. S ___ -- 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] Echo and static on PRI with errors
On Wed, Jul 1, 2009 at 5:09 AM, Tom O'Connor t...@twinhelix.org wrote: On Wed, Jul 1, 2009 at 7:37 AM, Francesco Peeters france...@fampeeters.com wrote: John F. Ervin wrote: What do you do if you find things sharing interrupts (IRQ 11) in my case with my X100P card. I believe there is some sort of internal audio card in my cheap slow PC. Check the BIOS whether you can: Change the IRQ assignments Disable the extra hardware using the same IRQ Or otherwise try changing the slot it is in... I had very good results in the past swapping card around Good luck! I did a bit of investigation WRT the IRQ settings on this box. 00:02.1 USB Controller: nVidia Corporation CK804 USB Controller (rev a3) (prog-if 20) Subsystem: Hewlett-Packard Company Device 3207 Flags: bus master, 66MHz, fast devsel, latency 0, IRQ 11 -- 01:05.0 VGA compatible controller: nVidia Corporation NV11 [GeForce2 MX/MX 400] (rev b2) Subsystem: Hewlett-Packard Company Device 3207 Flags: bus master, 66MHz, medium devsel, latency 64, IRQ 11 -- 02:00.0 Ethernet controller: Broadcom Corporation NetXtreme BCM5721 Gigabit Ethernet PCI Express (rev 11) Subsystem: Hewlett-Packard Company Device 3209 Flags: bus master, fast devsel, latency 0, IRQ 11 -- 81:01.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface Subsystem: Device 79fe:0001 Flags: bus master, medium devsel, latency 64, IRQ 11 So basically there's 2 network cards and a USB controller sharing IRQ 11 with the Openvox card. I wasn't able to find any settings in the bios to manually configure IRQ assignments :( Could someone tell me how to set which IRQ the ISDN card picks up? -- Tom O'Connor http://www.twinhelix.org t...@twinhelix.org No wonder you are having issues with everything on 11! If you cannot do it in BIOS, try moving the card to another slot as suggested. Disable everything you don't need. Do you need USB? Parallel port? Whatever, you get the picture, disable it. what do you get from cat /proc/interrupts? Maybe IRQ steering is something to look at as well? -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- 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] Fwd: Unknown udp ports listening experts calling !
On Wed, 2009-07-01 at 10:14 +0100, Steve Howes wrote: On 1 Jul 2009, at 09:54, Xavier Cardil wrote: udp0 0 0.0.0.0:2727 0.0.0.0:* 4989/asterisk udp0 0 0.0.0.0:9001 0.0.0.0:* 26354/udp-sender udp0 0 0.0.0.0:5000 0.0.0.0:* 4989/asterisk 2727 = mgcp I found that with Google. A useful tool. snip I thought 9001 was for JetDirect style print servers. I don't recall off the top of my head if they are tcp or udp - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- 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] Echo and static on PRI with errors
Tom O'Connor wrote: On Wed, Jul 1, 2009 at 7:37 AM, Francesco Peeters france...@fampeeters.com mailto:france...@fampeeters.com wrote: John F. Ervin wrote: What do you do if you find things sharing interrupts (IRQ 11) in my case with my X100P card. I believe there is some sort of internal audio card in my cheap slow PC. Check the BIOS whether you can: Change the IRQ assignments Disable the extra hardware using the same IRQ Or otherwise try changing the slot it is in... I had very good results in the past swapping card around Good luck! I did a bit of investigation WRT the IRQ settings on this box. 00:02.1 USB Controller: nVidia Corporation CK804 USB Controller (rev a3) (prog-if 20) Subsystem: Hewlett-Packard Company Device 3207 Flags: bus master, 66MHz, fast devsel, latency 0, IRQ 11 -- 01:05.0 VGA compatible controller: nVidia Corporation NV11 [GeForce2 MX/MX 400] (rev b2) Subsystem: Hewlett-Packard Company Device 3207 Flags: bus master, 66MHz, medium devsel, latency 64, IRQ 11 -- 02:00.0 Ethernet controller: Broadcom Corporation NetXtreme BCM5721 Gigabit Ethernet PCI Express (rev 11) Subsystem: Hewlett-Packard Company Device 3209 Flags: bus master, fast devsel, latency 0, IRQ 11 -- 81:01.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface Subsystem: Device 79fe:0001 Flags: bus master, medium devsel, latency 64, IRQ 11 So basically there's 2 network cards and a USB controller sharing IRQ 11 with the Openvox card. I wasn't able to find any settings in the bios to manually configure IRQ assignments :( Could someone tell me how to set which IRQ the ISDN card picks up? -- Tom O'Connor http://www.twinhelix.org t...@twinhelix.org mailto:t...@twinhelix.org Hi, Unfortunately is not always possible and it depends on how the mainboard was realized. For what I can understand a lot of producers decide to route only a subset of physical IRQ lines to the PCI slots (I think is something related to cost reduction) and to share it with other onboard peripherals. This lets impossible to change the IRQ assignment for expansion cards. This is not always true and sometimes swapping add-on cards solves the problem. We had better results with cards based on new Digium technology or with Sangoma cards. Best regards, Marco Signorini. ___ -- 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] Echo and static on PRI with errors
On Wed, Jul 1, 2009 at 10:49 AM, Marco Signorini marcota...@libero.itwrote: Tom O'Connor wrote: On Wed, Jul 1, 2009 at 7:37 AM, Francesco Peeters france...@fampeeters.com wrote: John F. Ervin wrote: What do you do if you find things sharing interrupts (IRQ 11) in my case with my X100P card. I believe there is some sort of internal audio card in my cheap slow PC. Check the BIOS whether you can: Change the IRQ assignments Disable the extra hardware using the same IRQ Or otherwise try changing the slot it is in... I had very good results in the past swapping card around Good luck! I did a bit of investigation WRT the IRQ settings on this box. 00:02.1 USB Controller: nVidia Corporation CK804 USB Controller (rev a3) (prog-if 20) Subsystem: Hewlett-Packard Company Device 3207 Flags: bus master, 66MHz, fast devsel, latency 0, IRQ 11 -- 01:05.0 VGA compatible controller: nVidia Corporation NV11 [GeForce2 MX/MX 400] (rev b2) Subsystem: Hewlett-Packard Company Device 3207 Flags: bus master, 66MHz, medium devsel, latency 64, IRQ 11 -- 02:00.0 Ethernet controller: Broadcom Corporation NetXtreme BCM5721 Gigabit Ethernet PCI Express (rev 11) Subsystem: Hewlett-Packard Company Device 3209 Flags: bus master, fast devsel, latency 0, IRQ 11 -- 81:01.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface Subsystem: Device 79fe:0001 Flags: bus master, medium devsel, latency 64, IRQ 11 So basically there's 2 network cards and a USB controller sharing IRQ 11 with the Openvox card. I wasn't able to find any settings in the bios to manually configure IRQ assignments :( Could someone tell me how to set which IRQ the ISDN card picks up? -- Tom O'Connor http://www.twinhelix.org t...@twinhelix.org Hi, Unfortunately is not always possible and it depends on how the mainboard was realized. For what I can understand a lot of producers decide to route only a subset of physical IRQ lines to the PCI slots (I think is something related to cost reduction) and to share it with other onboard peripherals. This lets impossible to change the IRQ assignment for expansion cards. This is not always true and sometimes swapping add-on cards solves the problem. We had better results with cards based on new Digium technology or with Sangoma cards. There is almost no room for manouvering in the HP bios. There's no ability to disable stuff like parallel ports, or anything else really. I don't think i'd buy digium hardware again. I'm already considering RMAing these cards and getting Sangoma ones. -- Tom O'Connor http://www.twinhelix.org t...@twinhelix.org ___ -- 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] Fwd: Unknown udp ports listening experts calling !
I found nothing is passing through those ports . . . I think something was sending the stream to our PST/SIP gateways, so the calls where affected when getting in to the gateways. I found we are not running any extra TCL applications on those gateways . . . could it be possible ? Could an UDP stream get mixed with another through an UDP port ? Is a very strange issue but I really want to know why . . . any more hints ? Thanks. On Wed, Jul 1, 2009 at 11:48 AM, John A. Sullivan III jsulli...@opensourcedevel.com wrote: On Wed, 2009-07-01 at 10:14 +0100, Steve Howes wrote: On 1 Jul 2009, at 09:54, Xavier Cardil wrote: udp0 0 0.0.0.0:2727 0.0.0.0:* 4989/asterisk udp0 0 0.0.0.0:9001 0.0.0.0:* 26354/udp-sender udp0 0 0.0.0.0:5000 0.0.0.0:* 4989/asterisk 2727 = mgcp I found that with Google. A useful tool. snip I thought 9001 was for JetDirect style print servers. I don't recall off the top of my head if they are tcp or udp - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- 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] Echo and static on PRI with errors
On Wed, Jul 1, 2009 at 5:58 AM, Tom O'Connor t...@twinhelix.org wrote: On Wed, Jul 1, 2009 at 10:49 AM, Marco Signorini marcota...@libero.itwrote: Tom O'Connor wrote: On Wed, Jul 1, 2009 at 7:37 AM, Francesco Peeters france...@fampeeters.com wrote: John F. Ervin wrote: What do you do if you find things sharing interrupts (IRQ 11) in my case with my X100P card. I believe there is some sort of internal audio card in my cheap slow PC. Check the BIOS whether you can: Change the IRQ assignments Disable the extra hardware using the same IRQ Or otherwise try changing the slot it is in... I had very good results in the past swapping card around Good luck! I did a bit of investigation WRT the IRQ settings on this box. 00:02.1 USB Controller: nVidia Corporation CK804 USB Controller (rev a3) (prog-if 20) Subsystem: Hewlett-Packard Company Device 3207 Flags: bus master, 66MHz, fast devsel, latency 0, IRQ 11 -- 01:05.0 VGA compatible controller: nVidia Corporation NV11 [GeForce2 MX/MX 400] (rev b2) Subsystem: Hewlett-Packard Company Device 3207 Flags: bus master, 66MHz, medium devsel, latency 64, IRQ 11 -- 02:00.0 Ethernet controller: Broadcom Corporation NetXtreme BCM5721 Gigabit Ethernet PCI Express (rev 11) Subsystem: Hewlett-Packard Company Device 3209 Flags: bus master, fast devsel, latency 0, IRQ 11 -- 81:01.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface Subsystem: Device 79fe:0001 Flags: bus master, medium devsel, latency 64, IRQ 11 So basically there's 2 network cards and a USB controller sharing IRQ 11 with the Openvox card. I wasn't able to find any settings in the bios to manually configure IRQ assignments :( Could someone tell me how to set which IRQ the ISDN card picks up? -- Tom O'Connor http://www.twinhelix.org t...@twinhelix.org Hi, Unfortunately is not always possible and it depends on how the mainboard was realized. For what I can understand a lot of producers decide to route only a subset of physical IRQ lines to the PCI slots (I think is something related to cost reduction) and to share it with other onboard peripherals. This lets impossible to change the IRQ assignment for expansion cards. This is not always true and sometimes swapping add-on cards solves the problem. We had better results with cards based on new Digium technology or with Sangoma cards. There is almost no room for manouvering in the HP bios. There's no ability to disable stuff like parallel ports, or anything else really. I don't think i'd buy digium hardware again. I'm already considering RMAing these cards and getting Sangoma ones. -- Tom O'Connor http://www.twinhelix.org t...@twinhelix.org That is one option. The new line Digium cards are on par with Sangoma as far as IRQ issues. I really like Sangoma's lifetime warranty though. I don't think Digium has countered that bold move. I would try the RMA and if that doesn't work, you can always pickup a decent last year's model server at http://www.surpluscomputers.com/featured-hardware/cg-69/servers.html For a basic asterisk server or PBX with nothing special going on, any of these servers are more than enough, even overkill. No affiliation, I have to say the shipping is high and they are slow to ship but the prices are great, never had an issue with any of their boxen (dozens, knock on wood) -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- 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] Echo and static on PRI with errors
On Wed, Jul 1, 2009 at 6:08 AM, Steve Totaro stot...@first-notification.com wrote: On Wed, Jul 1, 2009 at 5:58 AM, Tom O'Connor t...@twinhelix.org wrote: On Wed, Jul 1, 2009 at 10:49 AM, Marco Signorini marcota...@libero.itwrote: Tom O'Connor wrote: On Wed, Jul 1, 2009 at 7:37 AM, Francesco Peeters france...@fampeeters.com wrote: John F. Ervin wrote: What do you do if you find things sharing interrupts (IRQ 11) in my case with my X100P card. I believe there is some sort of internal audio card in my cheap slow PC. Check the BIOS whether you can: Change the IRQ assignments Disable the extra hardware using the same IRQ Or otherwise try changing the slot it is in... I had very good results in the past swapping card around Good luck! I did a bit of investigation WRT the IRQ settings on this box. 00:02.1 USB Controller: nVidia Corporation CK804 USB Controller (rev a3) (prog-if 20) Subsystem: Hewlett-Packard Company Device 3207 Flags: bus master, 66MHz, fast devsel, latency 0, IRQ 11 -- 01:05.0 VGA compatible controller: nVidia Corporation NV11 [GeForce2 MX/MX 400] (rev b2) Subsystem: Hewlett-Packard Company Device 3207 Flags: bus master, 66MHz, medium devsel, latency 64, IRQ 11 -- 02:00.0 Ethernet controller: Broadcom Corporation NetXtreme BCM5721 Gigabit Ethernet PCI Express (rev 11) Subsystem: Hewlett-Packard Company Device 3209 Flags: bus master, fast devsel, latency 0, IRQ 11 -- 81:01.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface Subsystem: Device 79fe:0001 Flags: bus master, medium devsel, latency 64, IRQ 11 So basically there's 2 network cards and a USB controller sharing IRQ 11 with the Openvox card. I wasn't able to find any settings in the bios to manually configure IRQ assignments :( Could someone tell me how to set which IRQ the ISDN card picks up? -- Tom O'Connor http://www.twinhelix.org t...@twinhelix.org Hi, Unfortunately is not always possible and it depends on how the mainboard was realized. For what I can understand a lot of producers decide to route only a subset of physical IRQ lines to the PCI slots (I think is something related to cost reduction) and to share it with other onboard peripherals. This lets impossible to change the IRQ assignment for expansion cards. This is not always true and sometimes swapping add-on cards solves the problem. We had better results with cards based on new Digium technology or with Sangoma cards. There is almost no room for manouvering in the HP bios. There's no ability to disable stuff like parallel ports, or anything else really. I don't think i'd buy digium hardware again. I'm already considering RMAing these cards and getting Sangoma ones. -- Tom O'Connor http://www.twinhelix.org t...@twinhelix.org That is one option. The new line Digium cards are on par with Sangoma as far as IRQ issues. I really like Sangoma's lifetime warranty though. I don't think Digium has countered that bold move. I would try the RMA and if that doesn't work, you can always pickup a decent last year's model server at http://www.surpluscomputers.com/featured-hardware/cg-69/servers.html For a basic asterisk server or PBX with nothing special going on, any of these servers are more than enough, even overkill. No affiliation, I have to say the shipping is high and they are slow to ship but the prices are great, never had an issue with any of their boxen (dozens, knock on wood) I wish I had seen this http://www.surpluscomputers.com/348694/ibm-10-pack-ibm-x335-dual.html before buying ten of these http://www.surpluscomputers.com/348663/hp-dl140-proliant-dual-xeon.html The ten pack of servers won't allow me to get past shipping so it may be a mistake but I am pretty sure it wasn't listed yesterday so maybe they just need to update shipping costs. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- 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] Echo and static on PRI with errors
On Wed, Jul 1, 2009 at 11:08 AM, Steve Totaro stot...@first-notification.com wrote: On Wed, Jul 1, 2009 at 5:58 AM, Tom O'Connor t...@twinhelix.org wrote: On Wed, Jul 1, 2009 at 10:49 AM, Marco Signorini marcota...@libero.itwrote: Tom O'Connor wrote: On Wed, Jul 1, 2009 at 7:37 AM, Francesco Peeters france...@fampeeters.com wrote: John F. Ervin wrote: What do you do if you find things sharing interrupts (IRQ 11) in my case with my X100P card. I believe there is some sort of internal audio card in my cheap slow PC. Check the BIOS whether you can: Change the IRQ assignments Disable the extra hardware using the same IRQ Or otherwise try changing the slot it is in... I had very good results in the past swapping card around Good luck! I did a bit of investigation WRT the IRQ settings on this box. 00:02.1 USB Controller: nVidia Corporation CK804 USB Controller (rev a3) (prog-if 20) Subsystem: Hewlett-Packard Company Device 3207 Flags: bus master, 66MHz, fast devsel, latency 0, IRQ 11 -- 01:05.0 VGA compatible controller: nVidia Corporation NV11 [GeForce2 MX/MX 400] (rev b2) Subsystem: Hewlett-Packard Company Device 3207 Flags: bus master, 66MHz, medium devsel, latency 64, IRQ 11 -- 02:00.0 Ethernet controller: Broadcom Corporation NetXtreme BCM5721 Gigabit Ethernet PCI Express (rev 11) Subsystem: Hewlett-Packard Company Device 3209 Flags: bus master, fast devsel, latency 0, IRQ 11 -- 81:01.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface Subsystem: Device 79fe:0001 Flags: bus master, medium devsel, latency 64, IRQ 11 So basically there's 2 network cards and a USB controller sharing IRQ 11 with the Openvox card. I wasn't able to find any settings in the bios to manually configure IRQ assignments :( Could someone tell me how to set which IRQ the ISDN card picks up? -- Tom O'Connor http://www.twinhelix.org t...@twinhelix.org Hi, Unfortunately is not always possible and it depends on how the mainboard was realized. For what I can understand a lot of producers decide to route only a subset of physical IRQ lines to the PCI slots (I think is something related to cost reduction) and to share it with other onboard peripherals. This lets impossible to change the IRQ assignment for expansion cards. This is not always true and sometimes swapping add-on cards solves the problem. We had better results with cards based on new Digium technology or with Sangoma cards. There is almost no room for manouvering in the HP bios. There's no ability to disable stuff like parallel ports, or anything else really. I don't think i'd buy digium hardware again. I'm already considering RMAing these cards and getting Sangoma ones. -- Tom O'Connor http://www.twinhelix.org t...@twinhelix.org That is one option. The new line Digium cards are on par with Sangoma as far as IRQ issues. I really like Sangoma's lifetime warranty though. I don't think Digium has countered that bold move. I would try the RMA and if that doesn't work, you can always pickup a decent last year's model server at http://www.surpluscomputers.com/featured-hardware/cg-69/servers.html For a basic asterisk server or PBX with nothing special going on, any of these servers are more than enough, even overkill. No affiliation, I have to say the shipping is high and they are slow to ship but the prices are great, never had an issue with any of their boxen (dozens, knock on wood) I don't really know what you mean about the new line Digium cards.. which models are in this new line? the server i'm using is hardly new, it's one of the older DL145s; so i don't think this would help much! I've tried swapping the card in the slots. no help :( -- Tom O'Connor http://www.twinhelix.org t...@twinhelix.org ___ -- 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] Fwd: Unknown udp ports listening experts calling !
Xavier Cardil wrote: I found nothing is passing through those ports . . . I think something was sending the stream to our PST/SIP gateways, so the calls where affected when getting in to the gateways. I found we are not running any extra TCL applications on those gateways . . . could it be possible ? Could an UDP stream get mixed with another through an UDP port ? Is a very strange issue but I really want to know why . . . any more hints ? Thanks. On Wed, Jul 1, 2009 at 11:48 AM, John A. Sullivan III jsulli...@opensourcedevel.com mailto:jsulli...@opensourcedevel.com wrote: On Wed, 2009-07-01 at 10:14 +0100, Steve Howes wrote: On 1 Jul 2009, at 09:54, Xavier Cardil wrote: udp0 0 0.0.0.0:2727 http://0.0.0.0:2727 0.0.0.0:* 4989/asterisk udp0 0 0.0.0.0:9001 http://0.0.0.0:9001 0.0.0.0:* 26354/udp-sender udp0 0 0.0.0.0:5000 http://0.0.0.0:5000 0.0.0.0:* 4989/asterisk 2727 = mgcp I found that with Google. A useful tool. snip I thought 9001 was for JetDirect style print servers. I don't recall off the top of my head if they are tcp or udp - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com mailto:jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society Assuming first your box doesn't have a rootkit installed (to check for a rootkit, use rkhunter. Your distro may have it packaged, if not google for it) I use lsof to find out what is listening to TCP and UDP ports: lsof -P | grep UDP lsof -P | grep TCP YMMV Bruce ___ -- 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] Welcome Message
You will need to insert the line before each place where you send calls
to Meetme and change the existing priority 1 to n. For example:
exten = 8600099,1,Playback(/var/lib/asterisk/sounds/silence/1)
exten = 8600099,n,Meetme(8600099)
exten = 8600100,1,Playback(/var/lib/asterisk/sounds/silence/1)
exten = 8600100,n,Meetme(8600100)
And so on...
This is assuming the path for sound files is:
/var/lib/asterisk/sounds/silence/1 You may need to modify the path if
your folder locations are different. Good luck!
- Josh
David @ULC wrote:
Thanks for the Reply,
I was waiting online for someone to reply : -)
Here is my Extension file : [ Where should I enter those line ? ]
exten = 8600099,1,Meetme(8600099)
exten = 8600100,1,Meetme(8600100)
exten = 8601,1,Meetme(8601)
exten = h,1,DeadAGI(agi://127.0.0.1:4577/call_log
http://127.0.0.1:4577/call_log)
exten =
h,2,DeadAGI(agi://127.0.0.1:4577/VD_hangup--HVcauses--PRI-NODEBUG-${HANGUPCAUSE}-${DIALSTATUS}-${DIALEDTIME}-${ANSWEREDTIME}
http://127.0.0.1:4577/VD_hangup--HVcauses--PRI-NODEBUG-$%7BHANGUPCAUSE%7D-$%7BDIALSTATUS%7D-$%7BDIALEDTIME%7D-$%7BANSWEREDTIME%7D))
exten = i,1,Playback(invalid)
exten = t,1,Goto(#,1)
exten = _68600XXX,1,Meetme(${EXTEN:1},mq)
exten = _78600XXX,1,Meetme(${EXTEN:1},q)
exten = _850266.,1,Wait(2)
exten = _850266.,2,Voicemail(${EXTEN:14})
exten = _850266.,3,Hangup()
exten = _851X,1,Answer()
exten = _851X,2,Playback(${EXTEN})
exten = _851X,3,Hangup()
exten = _90009.,1,Answer()
exten = _90009.,2,AGI(agi-VDADcloser.agi,${EXTEN}-START)
exten = _90009.,3,Hangup()
exten = _9X.,1,AGI(agi://127.0.0.1:4577/call_log
http://127.0.0.1:4577/call_log)
exten = _9X.,2,Dial(SIP/${EXTEN:1...@sip8||tTor)
exten = _9X.,3,Hangup()
exten = _8X.,1,AGI(agi://127.0.0.1:4577/call_log
http://127.0.0.1:4577/call_log)
exten = _8X.,2,Dial(SIP/${EXTEN:1...@sip209||tTor)
exten = _8X.,3,Hangup()
exten = _X38600XXX,1,MeetMeAdmin(${EXTEN:2},t,${EXTEN:0:1})
exten = _X38600XXX,2,Hangup()
exten = _X48600XXX,1,MeetMeAdmin(${EXTEN:2},T,${EXTEN:0:1})
exten = _X48600XXX,2,Hangup()
exten = _[1-7]X.,1,AGI(agi://127.0.0.1:4577/call_log
http://127.0.0.1:4577/call_log)
exten = _[1-7]X.,2,Dial(SIP/${ext...@sip8||tTor)
exten = _[1-7]X.,3,Hangup()
On Wed, Jul 1, 2009 at 6:19 AM, David @ULC ucoms2...@gmail.com
mailto:ucoms2...@gmail.com wrote:
When I login to the asterisk, I just hear the HALF of the welcome
message :
You are currently the instead of You are currently the only
person in the conference
Thats also, I hear it after 60 secs or so..
Asterisk 1.2.27
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Re: [asterisk-users] Fwd: Unknown udp ports listening experts calling !
Hi Bruce, thank you for your recommendations . . . I passed the test and the only wanrning is this one : /usr/sbin/unhide [ Warning ] /usr/sbin/useradd[ OK ] /usr/sbin/userdel[ OK ] /usr/sbin/usermod[ OK ] /usr/sbin/vipw [ OK ] /usr/sbin/unhide-linux26 [ Warning ] On Wed, Jul 1, 2009 at 1:42 PM, Bruce Ferrell bferr...@baywinds.org wrote: Xavier Cardil wrote: I found nothing is passing through those ports . . . I think something was sending the stream to our PST/SIP gateways, so the calls where affected when getting in to the gateways. I found we are not running any extra TCL applications on those gateways . . . could it be possible ? Could an UDP stream get mixed with another through an UDP port ? Is a very strange issue but I really want to know why . . . any more hints ? Thanks. On Wed, Jul 1, 2009 at 11:48 AM, John A. Sullivan III jsulli...@opensourcedevel.com mailto:jsulli...@opensourcedevel.com wrote: On Wed, 2009-07-01 at 10:14 +0100, Steve Howes wrote: On 1 Jul 2009, at 09:54, Xavier Cardil wrote: udp0 0 0.0.0.0:2727 http://0.0.0.0:2727 0.0.0.0:* 4989/asterisk udp0 0 0.0.0.0:9001 http://0.0.0.0:9001 0.0.0.0:* 26354/udp-sender udp0 0 0.0.0.0:5000 http://0.0.0.0:5000 0.0.0.0:* 4989/asterisk 2727 = mgcp I found that with Google. A useful tool. snip I thought 9001 was for JetDirect style print servers. I don't recall off the top of my head if they are tcp or udp - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com mailto:jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society Assuming first your box doesn't have a rootkit installed (to check for a rootkit, use rkhunter. Your distro may have it packaged, if not google for it) I use lsof to find out what is listening to TCP and UDP ports: lsof -P | grep UDP lsof -P | grep TCP YMMV Bruce ___ -- 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] Extension status as XML for an Aastra 57i
On Wed, Jul 1, 2009 at 1:10 AM, Olivieroza-4...@myamail.com wrote: The 57i phone has 6 soft buttons which can show the status of at least 16 phones (if you do not want to use the rest of the soft buttons which would give you another 16). Are you sure of that ? How can you set more than one single phone to light on or off a given BLF ? With a single button, I agree you can query more than one phone status but the associated light can't display more than one phone or am I missing something ? On the 57i, there are 6 soft buttons above and the screen, and 6 below. The top set can have up to 10 configurations, when you add more than 6, the bottom right button changes to Next.. and scrolls the screen over. The bottom can have up to 20, with the same next button. Each of these keys can be configured as BLF keys. -jonathan ___ -- 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] Multi-tenant parking broken in 1.6.1.1?
On Tue, Jun 30, 2009 at 11:53 PM, John A. Sullivan III jsulli...@opensourcedevel.com wrote: On Wed, 2009-07-01 at 02:17 -0400, John A. Sullivan III wrote: Hello, all. With the assistance of very helpful folks, our brand new multi-tenant setup seems to be working smoothly from start to finish with just a bump or two. The biggest is parking. Now that we got most kinks worked out, I'm a little more comfortable in trying to resolve this. There seem to be two problems: 1. Parking assigns parking spaces from the default group no matter what we do. I haven't tested this. 2. When the parked call timer expires, the callback to the original callee fails because a | delimiter is used in the Dial() function. This has been fixed in the 1.6.1 SVN, and you will have to back port a patch until these changes are rolled into another release. I was disappointed that more bug fixes were not included in 1.6.1.1. -Jonathan ___ -- 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] Multi-tenant parking broken in 1.6.1.1?
This has been fixed in the 1.6.1 SVN, and you will have to back port a patch until these changes are rolled into another release. I was disappointed that more bug fixes were not included in 1.6.1.1. -Jonathan Asterisk 1.6.1.1 was released for a security issue, AST-2009-001. Why would you think that more bug fixes would be in it? Security release are only supposed to have the fix for the issue that caused the release to take place. - Brad ___ -- 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] CRMy type app?
On 29/06/09 18:26, Gordon Henderson wrote: Looking for a (windows) app. that will listen to the manager interface then pop-up a web browser pointing to a page on an incoming phone call.. Not looking for outlook integration, or outbound dialling, just to recognise an incoming call and poke a URL at a website in a browser and I've absolutely no idea how to do it in the MS windows world... Any clues appreciated.. (More pointing to an existing app. rather than how to write it myself!) Hi Gordon, Have you looked at ADM before? It might be suitable... http://adm.hamnett.org/ Alan ___ -- 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] Multi-tenant parking broken in 1.6.1.1?
On Wed, 2009-07-01 at 07:17 -0700, Jonathan Thurman wrote: On Tue, Jun 30, 2009 at 11:53 PM, John A. Sullivan III jsulli...@opensourcedevel.com wrote: On Wed, 2009-07-01 at 02:17 -0400, John A. Sullivan III wrote: Hello, all. With the assistance of very helpful folks, our brand new multi-tenant setup seems to be working smoothly from start to finish with just a bump or two. The biggest is parking. Now that we got most kinks worked out, I'm a little more comfortable in trying to resolve this. There seem to be two problems: 1. Parking assigns parking spaces from the default group no matter what we do. I haven't tested this. 2. When the parked call timer expires, the callback to the original callee fails because a | delimiter is used in the Dial() function. This has been fixed in the 1.6.1 SVN, and you will have to back port a patch until these changes are rolled into another release. I was disappointed that more bug fixes were not included in 1.6.1.1. snip Phew! At least I know I'm not out of my mind! Being fairly new to the Asterisk community, which patch shall I look for and in what section of the SVN? Can I apply it to the release tarball (hopefully) or must I compile out of SVN (which I hate to do in a production environment)? Thanks very much - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- 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] Welcome Message
Any more suggestions ?
On Wed, Jul 1, 2009 at 8:30 AM, David @ULC ucoms2...@gmail.com wrote:
Thanks for the Reply,
I was waiting online for someone to reply : -)
Here is my Extension file : [ Where should I enter those line ? ]
exten = 8600099,1,Meetme(8600099)
exten = 8600100,1,Meetme(8600100)
exten = 8601,1,Meetme(8601)
exten = h,1,DeadAGI(agi://127.0.0.1:4577/call_log)
exten = h,2,DeadAGI(agi://
127.0.0.1:4577/VD_hangup--HVcauses--PRI-NODEBUG-${HANGUPCAUSE}-${DIALSTATUS}-${DIALEDTIME}-${ANSWEREDTIME}http://127.0.0.1:4577/VD_hangup--HVcauses--PRI-NODEBUG-$%7BHANGUPCAUSE%7D-$%7BDIALSTATUS%7D-$%7BDIALEDTIME%7D-$%7BANSWEREDTIME%7D
))
exten = i,1,Playback(invalid)
exten = t,1,Goto(#,1)
exten = _68600XXX,1,Meetme(${EXTEN:1},mq)
exten = _78600XXX,1,Meetme(${EXTEN:1},q)
exten = _850266.,1,Wait(2)
exten = _850266.,2,Voicemail(${EXTEN:14})
exten = _850266.,3,Hangup()
exten = _851X,1,Answer()
exten = _851X,2,Playback(${EXTEN})
exten = _851X,3,Hangup()
exten = _90009.,1,Answer()
exten = _90009.,2,AGI(agi-VDADcloser.agi,${EXTEN}-START)
exten = _90009.,3,Hangup()
exten = _9X.,1,AGI(agi://127.0.0.1:4577/call_log)
exten = _9X.,2,Dial(SIP/${EXTEN:1...@sip8||tTor)
exten = _9X.,3,Hangup()
exten = _8X.,1,AGI(agi://127.0.0.1:4577/call_log)
exten = _8X.,2,Dial(SIP/${EXTEN:1...@sip209||tTor)
exten = _8X.,3,Hangup()
exten = _X38600XXX,1,MeetMeAdmin(${EXTEN:2},t,${EXTEN:0:1})
exten = _X38600XXX,2,Hangup()
exten = _X48600XXX,1,MeetMeAdmin(${EXTEN:2},T,${EXTEN:0:1})
exten = _X48600XXX,2,Hangup()
exten = _[1-7]X.,1,AGI(agi://127.0.0.1:4577/call_log)
exten = _[1-7]X.,2,Dial(SIP/${ext...@sip8||tTor)
exten = _[1-7]X.,3,Hangup()
On Wed, Jul 1, 2009 at 6:19 AM, David @ULC ucoms2...@gmail.com wrote:
When I login to the asterisk, I just hear the HALF of the welcome message
:
You are currently the instead of You are currently the only person in
the conference
Thats also, I hear it after 60 secs or so..
Asterisk 1.2.27
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Re: [asterisk-users] Welcome Message
What was the result of my earlier suggestion? See below.
Joshua Billings wrote:
You will need to insert the line before each place where you send
calls to Meetme and change the existing priority 1 to n. For example:
exten = 8600099,1,Playback(/var/lib/asterisk/sounds/silence/1)
exten = 8600099,n,Meetme(8600099)
exten = 8600100,1,Playback(/var/lib/asterisk/sounds/silence/1)
exten = 8600100,n,Meetme(8600100)
And so on...
This is assuming the path for sound files is:
/var/lib/asterisk/sounds/silence/1 You may need to modify the path if
your folder locations are different. Good luck!
- Josh
David @ULC wrote:
Any more suggestions ?
On Wed, Jul 1, 2009 at 8:30 AM, David @ULC ucoms2...@gmail.com
mailto:ucoms2...@gmail.com wrote:
Thanks for the Reply,
I was waiting online for someone to reply : -)
Here is my Extension file : [ Where should I enter those line ? ]
exten = 8600099,1,Meetme(8600099)
exten = 8600100,1,Meetme(8600100)
exten = 8601,1,Meetme(8601)
exten = h,1,DeadAGI(agi://127.0.0.1:4577/call_log
http://127.0.0.1:4577/call_log)
exten =
h,2,DeadAGI(agi://127.0.0.1:4577/VD_hangup--HVcauses--PRI-NODEBUG-${HANGUPCAUSE}-${DIALSTATUS}-${DIALEDTIME}-${ANSWEREDTIME}
http://127.0.0.1:4577/VD_hangup--HVcauses--PRI-NODEBUG-$%7BHANGUPCAUSE%7D-$%7BDIALSTATUS%7D-$%7BDIALEDTIME%7D-$%7BANSWEREDTIME%7D))
exten = i,1,Playback(invalid)
exten = t,1,Goto(#,1)
exten = _68600XXX,1,Meetme(${EXTEN:1},mq)
exten = _78600XXX,1,Meetme(${EXTEN:1},q)
exten = _850266.,1,Wait(2)
exten = _850266.,2,Voicemail(${EXTEN:14})
exten = _850266.,3,Hangup()
exten = _851X,1,Answer()
exten = _851X,2,Playback(${EXTEN})
exten = _851X,3,Hangup()
exten = _90009.,1,Answer()
exten = _90009.,2,AGI(agi-VDADcloser.agi,${EXTEN}-START)
exten = _90009.,3,Hangup()
exten = _9X.,1,AGI(agi://127.0.0.1:4577/call_log
http://127.0.0.1:4577/call_log)
exten = _9X.,2,Dial(SIP/${EXTEN:1...@sip8||tTor)
exten = _9X.,3,Hangup()
exten = _8X.,1,AGI(agi://127.0.0.1:4577/call_log
http://127.0.0.1:4577/call_log)
exten = _8X.,2,Dial(SIP/${EXTEN:1...@sip209||tTor)
exten = _8X.,3,Hangup()
exten = _X38600XXX,1,MeetMeAdmin(${EXTEN:2},t,${EXTEN:0:1})
exten = _X38600XXX,2,Hangup()
exten = _X48600XXX,1,MeetMeAdmin(${EXTEN:2},T,${EXTEN:0:1})
exten = _X48600XXX,2,Hangup()
exten = _[1-7]X.,1,AGI(agi://127.0.0.1:4577/call_log
http://127.0.0.1:4577/call_log)
exten = _[1-7]X.,2,Dial(SIP/${ext...@sip8||tTor)
exten = _[1-7]X.,3,Hangup()
On Wed, Jul 1, 2009 at 6:19 AM, David @ULC ucoms2...@gmail.com
mailto:ucoms2...@gmail.com wrote:
When I login to the asterisk, I just hear the HALF of the
welcome message :
You are currently the instead of You are currently the
only person in the conference
Thats also, I hear it after 60 secs or so..
Asterisk 1.2.27
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[asterisk-users] * as VM for legacy PBX?
Hi, all. I've got an old Telrad PBX with an Emagen(?) voicemail box. The VM box, itself, is beginning to show its age. Big-time. We're thinking it might be time to look for a replacement. I'd love to install Asterisk with an FXO card or something, but I don't think it supports whatever protocol legacy PBX's used to speak to VM systems. If someone can tell me I'm wrong, a six pack of their favorite $BEVERAGE will magically appear at their door. Thanks much! -Ken -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Welcome Message
Have you verified that the sound file is intact (convert to wav with sox and
play thru mplayer, or just set up a test line exten =
7529,1,Playback(conf-onlyperson)?
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David @ULC
Sent: Wednesday, July 01, 2009 10:02 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Welcome Message
Any more suggestions ?
On Wed, Jul 1, 2009 at 8:30 AM, David @ULC ucoms2...@gmail.com wrote:
Thanks for the Reply,
I was waiting online for someone to reply : -)
Here is my Extension file : [ Where should I enter those line ? ]
exten = 8600099,1,Meetme(8600099)
exten = 8600100,1,Meetme(8600100)
exten = 8601,1,Meetme(8601)
exten = h,1,DeadAGI(agi://127.0.0.1:4577/call_log)
exten =
h,2,DeadAGI(agi://127.0.0.1:4577/VD_hangup--HVcauses--PRI-NODEBUG-${
HANGUPCAUSE}-${DIALSTATUS}-${DIALEDTIME}-${ANSWEREDTIME}
http://127.0.0.1:4577/VD_hangup--HVcauses--PRI-NODEBUG-$%7BHANGUPCA
USE%7D-$%7BDIALSTATUS%7D-$%7BDIALEDTIME%7D-$%7BANSWEREDTIME%7D
))
exten = i,1,Playback(invalid)
exten = t,1,Goto(#,1)
exten = _68600XXX,1,Meetme(${EXTEN:1},mq)
exten = _78600XXX,1,Meetme(${EXTEN:1},q)
exten = _850266.,1,Wait(2)
exten = _850266.,2,Voicemail(${EXTEN:14})
exten = _850266.,3,Hangup()
exten = _851X,1,Answer()
exten = _851X,2,Playback(${EXTEN})
exten = _851X,3,Hangup()
exten = _90009.,1,Answer()
exten = _90009.,2,AGI(agi-VDADcloser.agi,${EXTEN}-START)
exten = _90009.,3,Hangup()
exten = _9X.,1,AGI(agi://127.0.0.1:4577/call_log)
exten = _9X.,2,Dial(SIP/${EXTEN:1...@sip8||tTor)
exten = _9X.,3,Hangup()
exten = _8X.,1,AGI(agi://127.0.0.1:4577/call_log)
exten = _8X.,2,Dial(SIP/${EXTEN:1...@sip209||tTor)
exten = _8X.,3,Hangup()
exten = _X38600XXX,1,MeetMeAdmin(${EXTEN:2},t,${EXTEN:0:1})
exten = _X38600XXX,2,Hangup()
exten = _X48600XXX,1,MeetMeAdmin(${EXTEN:2},T,${EXTEN:0:1})
exten = _X48600XXX,2,Hangup()
exten = _[1-7]X.,1,AGI(agi://127.0.0.1:4577/call_log)
exten = _[1-7]X.,2,Dial(SIP/${ext...@sip8||tTor)
exten = _[1-7]X.,3,Hangup()
On Wed, Jul 1, 2009 at 6:19 AM, David @ULC ucoms2...@gmail.com wrote:
When I login to the asterisk, I just hear the HALF of the welcome message :
You are currently the instead of You are currently the only person in
the conference
Thats also, I hear it after 60 secs or so..
Asterisk 1.2.27
___
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Re: [asterisk-users] CRMy type app?
On Wed, 1 Jul 2009, Alan Lord (News) wrote: On 29/06/09 18:26, Gordon Henderson wrote: Looking for a (windows) app. that will listen to the manager interface then pop-up a web browser pointing to a page on an incoming phone call.. Not looking for outlook integration, or outbound dialling, just to recognise an incoming call and poke a URL at a website in a browser and I've absolutely no idea how to do it in the MS windows world... Any clues appreciated.. (More pointing to an existing app. rather than how to write it myself!) Hi Gordon, Have you looked at ADM before? It might be suitable... http://adm.hamnett.org/ I saw it - are you part of the team, or if not, then I hope someone from there is listening in... So I saw it, but you know what - the website didn't actually tell me what it does. I think it's great to have a bloggy/wordpressy/wiki sort of website, but the front page is lacking a missing What does ADM do paragraph or link to a page... Sure, there's screen shots, documentation, forums, etc. but if there was a single paragraph at the top that said exactly what it can do, then I'd have spent more time looking at it.. I have now spent some time on the site, but since I've already tested ADAT and it does what I need, it'll take some persuading to make me change... 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] Multi-tenant parking broken in 1.6.1.1?
This has been fixed in the 1.6.1 SVN, and you will have to back port a patch until these changes are rolled into another release. I was disappointed that more bug fixes were not included in 1.6.1.1. -Jonathan Asterisk 1.6.1.1 was released for a security issue, AST-2009-001. Why would you think that more bug fixes would be in it? Security release are only supposed to have the fix for the issue that caused the release to take place. - Brad Sorry, I am relatively new to the Asterisk project and probably don't fully understand how the release cycle for this project works. Are you saying that the minor releases are only for security bugs? I haven't seen anything in the on-line documentation that states this. I would think that major usability issues (like parked calls getting dropped if you don't pick them up) would be addressed in a release, not only in SVN. To me the point of a minor release is to fix bugs. It is sometimes quite a headache to download the latest release, have an issue, dig through the issue tracker to find that it was fixed a month ago, then update to SVN or back port a patch. This is especially difficult for those that are new to the project/community. -Jonathan ___ -- 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] * as VM for legacy PBX?
On Wed, Jul 1, 2009 at 11:16 AM, Ken D'Ambrosio k...@jots.org wrote: Hi, all. I've got an old Telrad PBX with an Emagen(?) voicemail box. The VM box, itself, is beginning to show its age. Big-time. We're thinking it might be time to look for a replacement. I'd love to install Asterisk with an FXO card or something, but I don't think it supports whatever protocol legacy PBX's used to speak to VM systems. If someone can tell me I'm wrong, a six pack of their favorite $BEVERAGE will magically appear at their door. Thanks much! -Ken I have done this many times. First, unplug one of the ports that goes to your voicemail and plug a regular pots phone into it (PBX). Make a call to VM (has to go out on the port you have the handset plugged into), answer it and listen. If you hear a bunch of DTMF then you are golden. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Welcome Message
*
*
*/var/lib/asterisk/sounds/silence/1*
*
*
*1 is the folder or the filename ?*
On Wed, Jul 1, 2009 at 8:30 AM, David @ULC ucoms2...@gmail.com wrote:
Thanks for the Reply,
I was waiting online for someone to reply : -)
Here is my Extension file : [ Where should I enter those line ? ]
exten = 8600099,1,Meetme(8600099)
exten = 8600100,1,Meetme(8600100)
exten = 8601,1,Meetme(8601)
exten = h,1,DeadAGI(agi://127.0.0.1:4577/call_log)
exten = h,2,DeadAGI(agi://
127.0.0.1:4577/VD_hangup--HVcauses--PRI-NODEBUG-${HANGUPCAUSE}-${DIALSTATUS}-${DIALEDTIME}-${ANSWEREDTIME}http://127.0.0.1:4577/VD_hangup--HVcauses--PRI-NODEBUG-$%7BHANGUPCAUSE%7D-$%7BDIALSTATUS%7D-$%7BDIALEDTIME%7D-$%7BANSWEREDTIME%7D
))
exten = i,1,Playback(invalid)
exten = t,1,Goto(#,1)
exten = _68600XXX,1,Meetme(${EXTEN:1},mq)
exten = _78600XXX,1,Meetme(${EXTEN:1},q)
exten = _850266.,1,Wait(2)
exten = _850266.,2,Voicemail(${EXTEN:14})
exten = _850266.,3,Hangup()
exten = _851X,1,Answer()
exten = _851X,2,Playback(${EXTEN})
exten = _851X,3,Hangup()
exten = _90009.,1,Answer()
exten = _90009.,2,AGI(agi-VDADcloser.agi,${EXTEN}-START)
exten = _90009.,3,Hangup()
exten = _9X.,1,AGI(agi://127.0.0.1:4577/call_log)
exten = _9X.,2,Dial(SIP/${EXTEN:1...@sip8||tTor)
exten = _9X.,3,Hangup()
exten = _8X.,1,AGI(agi://127.0.0.1:4577/call_log)
exten = _8X.,2,Dial(SIP/${EXTEN:1...@sip209||tTor)
exten = _8X.,3,Hangup()
exten = _X38600XXX,1,MeetMeAdmin(${EXTEN:2},t,${EXTEN:0:1})
exten = _X38600XXX,2,Hangup()
exten = _X48600XXX,1,MeetMeAdmin(${EXTEN:2},T,${EXTEN:0:1})
exten = _X48600XXX,2,Hangup()
exten = _[1-7]X.,1,AGI(agi://127.0.0.1:4577/call_log)
exten = _[1-7]X.,2,Dial(SIP/${ext...@sip8||tTor)
exten = _[1-7]X.,3,Hangup()
On Wed, Jul 1, 2009 at 6:19 AM, David @ULC ucoms2...@gmail.com wrote:
When I login to the asterisk, I just hear the HALF of the welcome message
:
You are currently the instead of You are currently the only person in
the conference
Thats also, I hear it after 60 secs or so..
Asterisk 1.2.27
___
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Re: [asterisk-users] Welcome Message
sound file is intact
Yes. I checked it with my other server.
On Wed, Jul 1, 2009 at 9:14 PM, David @ULC ucoms2...@gmail.com wrote:
*
*
*/var/lib/asterisk/sounds/silence/1*
*
*
*1 is the folder or the filename ?*
On Wed, Jul 1, 2009 at 8:30 AM, David @ULC ucoms2...@gmail.com wrote:
Thanks for the Reply,
I was waiting online for someone to reply : -)
Here is my Extension file : [ Where should I enter those line ? ]
exten = 8600099,1,Meetme(8600099)
exten = 8600100,1,Meetme(8600100)
exten = 8601,1,Meetme(8601)
exten = h,1,DeadAGI(agi://127.0.0.1:4577/call_log)
exten = h,2,DeadAGI(agi://
127.0.0.1:4577/VD_hangup--HVcauses--PRI-NODEBUG-${HANGUPCAUSE}-${DIALSTATUS}-${DIALEDTIME}-${ANSWEREDTIME}http://127.0.0.1:4577/VD_hangup--HVcauses--PRI-NODEBUG-$%7BHANGUPCAUSE%7D-$%7BDIALSTATUS%7D-$%7BDIALEDTIME%7D-$%7BANSWEREDTIME%7D
))
exten = i,1,Playback(invalid)
exten = t,1,Goto(#,1)
exten = _68600XXX,1,Meetme(${EXTEN:1},mq)
exten = _78600XXX,1,Meetme(${EXTEN:1},q)
exten = _850266.,1,Wait(2)
exten = _850266.,2,Voicemail(${EXTEN:14})
exten = _850266.,3,Hangup()
exten = _851X,1,Answer()
exten = _851X,2,Playback(${EXTEN})
exten = _851X,3,Hangup()
exten = _90009.,1,Answer()
exten = _90009.,2,AGI(agi-VDADcloser.agi,${EXTEN}-START)
exten = _90009.,3,Hangup()
exten = _9X.,1,AGI(agi://127.0.0.1:4577/call_log)
exten = _9X.,2,Dial(SIP/${EXTEN:1...@sip8||tTor)
exten = _9X.,3,Hangup()
exten = _8X.,1,AGI(agi://127.0.0.1:4577/call_log)
exten = _8X.,2,Dial(SIP/${EXTEN:1...@sip209||tTor)
exten = _8X.,3,Hangup()
exten = _X38600XXX,1,MeetMeAdmin(${EXTEN:2},t,${EXTEN:0:1})
exten = _X38600XXX,2,Hangup()
exten = _X48600XXX,1,MeetMeAdmin(${EXTEN:2},T,${EXTEN:0:1})
exten = _X48600XXX,2,Hangup()
exten = _[1-7]X.,1,AGI(agi://127.0.0.1:4577/call_log)
exten = _[1-7]X.,2,Dial(SIP/${ext...@sip8||tTor)
exten = _[1-7]X.,3,Hangup()
On Wed, Jul 1, 2009 at 6:19 AM, David @ULC ucoms2...@gmail.com wrote:
When I login to the asterisk, I just hear the HALF of the welcome message
:
You are currently the instead of You are currently the only person in
the conference
Thats also, I hear it after 60 secs or so..
Asterisk 1.2.27
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Re: [asterisk-users] Welcome Message
1 is the filename. The Playback application does not require you to
specify the extension. The idea is that by playing 1 second of silence
the message for MeetMe remains intact. Let me know how it goes.
- Josh
David @ULC wrote:
/
/
//var/lib/asterisk/sounds/silence/1/
/
/
/1 is the folder or the filename ?/
On Wed, Jul 1, 2009 at 8:30 AM, David @ULC ucoms2...@gmail.com
mailto:ucoms2...@gmail.com wrote:
Thanks for the Reply,
I was waiting online for someone to reply : -)
Here is my Extension file : [ Where should I enter those line ? ]
exten = 8600099,1,Meetme(8600099)
exten = 8600100,1,Meetme(8600100)
exten = 8601,1,Meetme(8601)
exten = h,1,DeadAGI(agi://127.0.0.1:4577/call_log
http://127.0.0.1:4577/call_log)
exten =
h,2,DeadAGI(agi://127.0.0.1:4577/VD_hangup--HVcauses--PRI-NODEBUG-${HANGUPCAUSE}-${DIALSTATUS}-${DIALEDTIME}-${ANSWEREDTIME}
http://127.0.0.1:4577/VD_hangup--HVcauses--PRI-NODEBUG-$%7BHANGUPCAUSE%7D-$%7BDIALSTATUS%7D-$%7BDIALEDTIME%7D-$%7BANSWEREDTIME%7D))
exten = i,1,Playback(invalid)
exten = t,1,Goto(#,1)
exten = _68600XXX,1,Meetme(${EXTEN:1},mq)
exten = _78600XXX,1,Meetme(${EXTEN:1},q)
exten = _850266.,1,Wait(2)
exten = _850266.,2,Voicemail(${EXTEN:14})
exten = _850266.,3,Hangup()
exten = _851X,1,Answer()
exten = _851X,2,Playback(${EXTEN})
exten = _851X,3,Hangup()
exten = _90009.,1,Answer()
exten = _90009.,2,AGI(agi-VDADcloser.agi,${EXTEN}-START)
exten = _90009.,3,Hangup()
exten = _9X.,1,AGI(agi://127.0.0.1:4577/call_log
http://127.0.0.1:4577/call_log)
exten = _9X.,2,Dial(SIP/${EXTEN:1...@sip8||tTor)
exten = _9X.,3,Hangup()
exten = _8X.,1,AGI(agi://127.0.0.1:4577/call_log
http://127.0.0.1:4577/call_log)
exten = _8X.,2,Dial(SIP/${EXTEN:1...@sip209||tTor)
exten = _8X.,3,Hangup()
exten = _X38600XXX,1,MeetMeAdmin(${EXTEN:2},t,${EXTEN:0:1})
exten = _X38600XXX,2,Hangup()
exten = _X48600XXX,1,MeetMeAdmin(${EXTEN:2},T,${EXTEN:0:1})
exten = _X48600XXX,2,Hangup()
exten = _[1-7]X.,1,AGI(agi://127.0.0.1:4577/call_log
http://127.0.0.1:4577/call_log)
exten = _[1-7]X.,2,Dial(SIP/${ext...@sip8||tTor)
exten = _[1-7]X.,3,Hangup()
On Wed, Jul 1, 2009 at 6:19 AM, David @ULC ucoms2...@gmail.com
mailto:ucoms2...@gmail.com wrote:
When I login to the asterisk, I just hear the HALF of the
welcome message :
You are currently the instead of You are currently the
only person in the conference
Thats also, I hear it after 60 secs or so..
Asterisk 1.2.27
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Re: [asterisk-users] Echo and static on PRI with errors
Could someone tell me how to set which IRQ the ISDN card picks up? It's a multi-stage process. Each PCI slot has four interrupt pins: INTA through INTD. A PCI card can choose to use any of these four (or even more than one of them, as some multi-port serial cards do). Most PCI cards use only one pin: usually INTA. The motherboard routes four interrupt lines between the pins in the slots it provides. The motherboard usually does *not* route a line to the same pin on all slots... for example, INTA on slot 1 might be routed to INTB on slot 2 and INTC on slot 3, and then back to INTA on slot 4. This mix 'em up routing is done to help compensate for the fact that most PCI cards use only INTA - it keeps all the cards from pounding on the same interrupt line. This is also why one way to move a PCI card to a different IRQ, is to move it to a different slot. The motherboard must then route the interrupt lines to one or more IRQs. On classic PCI motherboards, with traditional PC interrupt controllers, there are only a very limited number of IRQs available (up through IRQ15) and many of these IRQs have dedicated functions and cannot be shared (e.g. any IRQ assigned to an ISA device can't be shared). As a result, these motherboards tend to route multiple PCI interrupts to only one or two IRQs - as in your case, where a whole boatload of things are being routed to IRQ11. On these traditional motherboards, all of the IRQ routing is under the control of the BIOS. Hence, the second way to un-burden IRQ11 would be to change your BIOS settings (as previously suggested). You would want to disable any unused devices - in particular, any IRQ-using ISA devices such as the parallel and serial ports - and mark these IRQs as available, not reserved for ISA. A good BIOS would then change the PCI-INT-to-IRQ routing and spread out the interrupt load. Unfortunately, it sounds as if the HP BIOS is of the Father Knows Best variety, and won't let you control your settings. Unless you can find an expert menu, or a separate configuration program for the BIOS data (sometimes vendors make a DOS or self-booting program available, rather than putting the full BIOS configuration in the BIOS itself) you're stuck here. There's a third possibility: APIC, the Advanced Programmable Interrupt Controller. This is a newer interrupt-controller architecture, present on SMP systems and on many modern uniprocessor systems. It provides the hardware and the OS with much more flexibility, and with quite a few additional IRQ numbers not supported by the traditional controller. You could try building a custom Linux kernel for your system, using a current stable kernel version (a 2.6 spin, at the moment). Enable APIC support, including the APIC on uniprocessor and local APIC support features. Boot this kernel, do an lspci -v, and see where your various cards and devices end up IRQ'ing. You may find that the APIC support has allowed the kernel to map these devices onto a wider range of IRQ numbers than previously. Unfortunately even this approach may not help on some motherboards. If the vendor has wired all of the INTA pins on the slots to the same line, and has also used this same line for the interrupts from the internal (non-slotted) PCI devices, then you'd be completely out of luck - it would be physically impossible to distribute the interrupts from these devices to different IRQs. My guess is that your biggest conflict is between the PRI card, and the network interfaces, since both are likely to be generators of lots of interrupts. Ugh... I just noticed something else... it looks as if the motherboard in question is using at least one PCI-to-PCI bridge: 81:01.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface Note the bus number: 81 hex. I think that this means that the card is sitting on the far side of a bridge chip... these are often used if a system has more PCI devices or slots than a single bus can support. I've had some bad experiences with bridged PCI systems in the past - some bridge chips seem to add quite a bit of latency to PCI bus access, or reduce bus throughput by quite a lot. Apparently the individual read and write transactions through the bridge suffer from a significant per-transaction overhead. I wonder whether IRQ latency/delay might not also be a problem here, or whether the bridge architecture might be forcing interrupts from some cards to use a single line/IRQ. ___ -- 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] Welcome Message
*This is what I get when I reload in CLI :* == Parsing '/etc/asterisk/extconfig.conf': Found == Parsing '/etc/asterisk/manager.conf': Found == Parsing '/etc/asterisk/cdr.conf': Found Jul 1 12:05:59 NOTICE[23347]: cdr.c:1214 do_reload: CDR simple logging enabled. == Parsing '/etc/asterisk/enum.conf': Found == Parsing '/etc/asterisk/rtp.conf': Found == RTP Allocating from port range 1 - 2 == Parsing '/etc/asterisk/dnsmgr.conf': Found Jul 1 12:05:59 NOTICE[23347]: dnsmgr.c:338 do_reload: Managed DNS entries will be refreshed every 300 seconds. -- Reloading module 'res_musiconhold.so' (Music On Hold Resource) == Parsing '/etc/asterisk/musiconhold.conf': Found -- Reloading module 'res_adsi.so' (ADSI Resource) == Parsing '/etc/asterisk/adsi.conf': Found -- Reloading module 'res_features.so' (Call Features Resource) == Parsing '/etc/asterisk/features.conf': Found -- Added extension '700' priority 1 to parkedcalls -- Reloading module 'res_crypto.so' (Cryptographic Digital Signatures) -- Reloading module 'res_indications.so' (Indications Configuration) -- Unregistered indication country 'at' -- Unregistered indication country 'au' -- Unregistered indication country 'br' -- Unregistered indication country 'be' -- Unregistered indication country 'ch' -- Unregistered indication country 'cl' -- Unregistered indication country 'cn' -- Unregistered indication country 'cz' -- Unregistered indication country 'de' -- Unregistered indication country 'dk' -- Unregistered indication country 'ee' -- Unregistered indication country 'es' -- Unregistered indication country 'fi' -- Unregistered indication country 'fr' -- Unregistered indication country 'gr' -- Unregistered indication country 'hu' -- Unregistered indication country 'it' -- Unregistered indication country 'lt' -- Unregistered indication country 'mx' -- Unregistered indication country 'nl' -- Unregistered indication country 'no' -- Unregistered indication country 'nz' -- Unregistered indication country 'pl' -- Unregistered indication country 'pt' -- Unregistered indication country 'ru' -- Unregistered indication country 'se' -- Unregistered indication country 'sg' -- Unregistered indication country 'uk' Jul 1 12:05:59 NOTICE[23347]: indications.c:505 ast_unregister_indication_country: Removed default indication country 'us' -- Unregistered indication country 'us' -- Unregistered indication country 'us-o' -- Unregistered indication country 'tw' -- Unregistered indication country 'za' == Parsing '/etc/asterisk/indications.conf': Found -- Registered indication country 'at' -- Registered indication country 'au' -- Registered indication country 'br' -- Registered indication country 'be' -- Registered indication country 'ch' -- Registered indication country 'cl' -- Registered indication country 'cn' -- Registered indication country 'cz' -- Registered indication country 'de' -- Registered indication country 'dk' -- Registered indication country 'ee' -- Registered indication country 'es' -- Registered indication country 'fi' -- Registered indication country 'fr' -- Registered indication country 'gr' -- Registered indication country 'hu' -- Registered indication country 'it' -- Registered indication country 'lt' -- Registered indication country 'mx' -- Registered indication country 'nl' -- Registered indication country 'no' -- Registered indication country 'nz' -- Registered indication country 'pl' -- Registered indication country 'pt' -- Registered indication country 'ru' -- Registered indication country 'se' -- Registered indication country 'sg' -- Registered indication country 'uk' -- Registered indication country 'us' -- Registered indication country 'us-o' -- Registered indication country 'tw' -- Registered indication country 'za' -- Setting default indication country to 'us' -- Reloading module 'pbx_config.so' (Text Extension Configuration) == Parsing '/etc/asterisk/extensions.conf': Found Jul 1 12:05:59 WARNING[23347]: pbx.c:3783 ast_merge_contexts_and_delete: Requested contexts didn't get merged -- Reloading module 'pbx_dundi.so' (Distributed Universal Number Discovery (DUNDi)) == Parsing '/etc/asterisk/dundi.conf': Found -- Reloading module 'pbx_ael.so' (Asterisk Extension Language Compiler) -- Registered extension context 'macro-std-exten-ael' -- Added extension 's' priority 1 to macro-std-exten-ael -- Added extension 's' priority 2 to macro-std-exten-ael -- Added extension 's' priority 3 to macro-std-exten-ael -- Added extension 's' priority 4 to macro-std-exten-ael -- Added extension 's' priority 5 to macro-std-exten-ael -- Added extension 'sw-4-BUSY' priority 1 to macro-std-exten-ael -- Added extension
Re: [asterisk-users] * as VM for legacy PBX?
On Wed, 1 Jul 2009, Ken D'Ambrosio wrote: Hi, all. I've got an old Telrad PBX with an Emagen(?) voicemail box. The VM box, itself, is beginning to show its age. Big-time. We're thinking it might be time to look for a replacement. I'd love to install Asterisk with an FXO card or something, but I don't think it supports whatever protocol legacy PBX's used to speak to VM systems. If someone can tell me I'm wrong, a six pack of their favorite $BEVERAGE will magically appear at their door. Thanks much! -Ken Why keep any of it? Migrate the whole thing to Asterisk. Usually old VM interfaces are nothing more than analog extensions, anyway, so your original plan is probably solid. j -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ -- 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] CRMy type app?
On 01/07/09 16:29, Gordon Henderson wrote: On Wed, 1 Jul 2009, Alan Lord (News) wrote: On 29/06/09 18:26, Gordon Henderson wrote: Looking for a (windows) app. that will listen to the manager interface then pop-up a web browser pointing to a page on an incoming phone call.. Not looking for outlook integration, or outbound dialling, just to recognise an incoming call and poke a URL at a website in a browser and I've absolutely no idea how to do it in the MS windows world... Any clues appreciated.. (More pointing to an existing app. rather than how to write it myself!) Hi Gordon, Have you looked at ADM before? It might be suitable... http://adm.hamnett.org/ I saw it - are you part of the team, or if not, then I hope someone from there is listening in... So I saw it, but you know what - the website didn't actually tell me what it does. I think it's great to have a bloggy/wordpressy/wiki sort of website, but the front page is lacking a missing What does ADM do paragraph or link to a page... Sure, there's screen shots, documentation, forums, etc. but if there was a single paragraph at the top that said exactly what it can do, then I'd have spent more time looking at it.. I have now spent some time on the site, but since I've already tested ADAT and it does what I need, it'll take some persuading to make me change... lol, I'm not anything to do with it. My business partner is using it though. It's a Java app (cross platform) that does the things you were looking for. It does seem to work for him although I haven't tried it myself. I just saw your post and thought it seemed to be a good fit. From an earlier blog post on his site: ADM provides some great features: * Automatic on-call volume reduction * One click dial from clipboard (paste number onto tray icon) * Integrated phonebook * List/Redirect/Hangup all active calls * One click call forward setup * Bluetooth presence detection to redirect calls when you walk out of the office * Pop up browser on incoming call (integrate with your CRM to auto load customers details when they call) * Cisco phone integration (auto speakerphone) * Slide-in popup on incoming call, with Answer(cisco only), Hold, Busy and Redirect buttons , CallerID and duration Cheers Al PS - he's a Brit too. ___ -- 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] Echo and static on PRI with errors
Dave Platt wrote: Could someone tell me how to set which IRQ the ISDN card picks up? It's a multi-stage process. Each PCI slot has four interrupt pins: INTA through INTD. A PCI card can choose to use any of these four (or even more than .. bridge architecture might be forcing interrupts from some cards to use a single line/IRQ. Thank you for your complete description on how PCI IRQ subsystem works. It's probably the best explanation I've found since years. My warm compliments, you've my best appreciation. Regards, Marco Signorini. Ingegni TECH S.r.l. http://www.ingegnitech.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] * as VM for legacy PBX?
As others have said, this is certainly possible. Our old NEC phone system had us in the same boat. It triggered voicemail by ringing the VM extension(s) and sending a DTMF burst of the extension to record VM for within 1.5 seconds. In our case, when any call came it in went to the voicemail system to play the main menu and allow the person to dial an extension. With that we were able to move a small set of power users to SIP phones for testing before we decided on a final phone and moved the whole campus. Daniel On Jul 1, 2009, at 8:16 AM, Ken D'Ambrosio wrote: Hi, all. I've got an old Telrad PBX with an Emagen(?) voicemail box. The VM box, itself, is beginning to show its age. Big-time. We're thinking it might be time to look for a replacement. I'd love to install Asterisk with an FXO card or something, but I don't think it supports whatever protocol legacy PBX's used to speak to VM systems. If someone can tell me I'm wrong, a six pack of their favorite $BEVERAGE will magically appear at their door. Thanks much! -Ken -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ -- 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] * as VM for legacy PBX?
Make a call to VM (has to go out on the port you have the handset plugged into), answer it and listen. If you hear a bunch of DTMF then you are golden. Sounds like good stuff, but my most substantial concerns involved things like MWI: is asterisk able to push that back to the PBX? -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ -- 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 message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ -- 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] Testing the manager.conf: sending and receiving commands
Hi All; How can I test manager.conf? Can I telnet to the asterisk machine at the port 5038 and send and receive commands to test if the manager is working fine? How? Regards Bilal ___ -- 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] * as VM for legacy PBX?
On Wed, Jul 1, 2009 at 12:15 PM, Jeff LaCoursiere j...@jeff.net wrote: On Wed, 1 Jul 2009, Ken D'Ambrosio wrote: Hi, all. I've got an old Telrad PBX with an Emagen(?) voicemail box. The VM box, itself, is beginning to show its age. Big-time. We're thinking it might be time to look for a replacement. I'd love to install Asterisk with an FXO card or something, but I don't think it supports whatever protocol legacy PBX's used to speak to VM systems. If someone can tell me I'm wrong, a six pack of their favorite $BEVERAGE will magically appear at their door. Thanks much! -Ken Why keep any of it? Migrate the whole thing to Asterisk. Usually old VM interfaces are nothing more than analog extensions, anyway, so your original plan is probably solid. I have also put Asterisk in front of a proprietary system to get all the functionality of Asterisk and save a fortune on a fork lift upgrade. Some people don't have the time or money, especially right now If the phone system works fine, why replace it? If it ain't broke, don't fix it If the users like it and it works for them, why force change. You can still get meetme, VM to email, so what is the big deal? Now if the VM system is heading for failure, Asterisk is a great replacement at a tiny fraction of what a replacement proprietary system will cost, even used. I have over a dozen (happy) customers with some sort of hybrid Asterisk/Proprietary setup. In addition, you can slowly migrate over to Asterisk if you want to, when you want to. With some dialplan magic, you can have half the company on the Telrad and the other on Asterisk and switch them over one by one. I cannot think of a better migration plan, if that is ever even in the plans. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- 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] Echo and static on PRI with errors
- Marco Signorini marcota...@libero.it wrote: Dave Platt wrote: Could someone tell me how to set which IRQ the ISDN card picks up? It's a multi-stage process. Each PCI slot has four interrupt pins: INTA through INTD. A PCI card can choose to use any of these four (or even more than .. bridge architecture might be forcing interrupts from some cards to use a single line/IRQ. Thank you for your complete description on how PCI IRQ subsystem works. It's probably the best explanation I've found since years. My warm compliments, you've my best appreciation. Regards, Marco Signorini. Agreed! That was a most thorough explanation which is greatly helpful. Thanks for posting it! --Tim ___ -- 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] Echo and static on PRI with errors
On Wed, Jul 1, 2009 at 6:35 AM, Tom O'Connor t...@twinhelix.org wrote: On Wed, Jul 1, 2009 at 11:08 AM, Steve Totaro stot...@first-notification.com wrote: On Wed, Jul 1, 2009 at 5:58 AM, Tom O'Connor t...@twinhelix.org wrote: On Wed, Jul 1, 2009 at 10:49 AM, Marco Signorini marcota...@libero.it wrote: Tom O'Connor wrote: On Wed, Jul 1, 2009 at 7:37 AM, Francesco Peeters france...@fampeeters.com wrote: John F. Ervin wrote: What do you do if you find things sharing interrupts (IRQ 11) in my case with my X100P card. I believe there is some sort of internal audio card in my cheap slow PC. Check the BIOS whether you can: Change the IRQ assignments Disable the extra hardware using the same IRQ Or otherwise try changing the slot it is in... I had very good results in the past swapping card around Good luck! I did a bit of investigation WRT the IRQ settings on this box. 00:02.1 USB Controller: nVidia Corporation CK804 USB Controller (rev a3) (prog-if 20) Subsystem: Hewlett-Packard Company Device 3207 Flags: bus master, 66MHz, fast devsel, latency 0, IRQ 11 -- 01:05.0 VGA compatible controller: nVidia Corporation NV11 [GeForce2 MX/MX 400] (rev b2) Subsystem: Hewlett-Packard Company Device 3207 Flags: bus master, 66MHz, medium devsel, latency 64, IRQ 11 -- 02:00.0 Ethernet controller: Broadcom Corporation NetXtreme BCM5721 Gigabit Ethernet PCI Express (rev 11) Subsystem: Hewlett-Packard Company Device 3209 Flags: bus master, fast devsel, latency 0, IRQ 11 -- 81:01.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface Subsystem: Device 79fe:0001 Flags: bus master, medium devsel, latency 64, IRQ 11 So basically there's 2 network cards and a USB controller sharing IRQ 11 with the Openvox card. I wasn't able to find any settings in the bios to manually configure IRQ assignments :( Could someone tell me how to set which IRQ the ISDN card picks up? -- Tom O'Connor http://www.twinhelix.org t...@twinhelix.org Hi, Unfortunately is not always possible and it depends on how the mainboard was realized. For what I can understand a lot of producers decide to route only a subset of physical IRQ lines to the PCI slots (I think is something related to cost reduction) and to share it with other onboard peripherals. This lets impossible to change the IRQ assignment for expansion cards. This is not always true and sometimes swapping add-on cards solves the problem. We had better results with cards based on new Digium technology or with Sangoma cards. There is almost no room for manouvering in the HP bios. There's no ability to disable stuff like parallel ports, or anything else really. I don't think i'd buy digium hardware again. I'm already considering RMAing these cards and getting Sangoma ones. -- Tom O'Connor http://www.twinhelix.org t...@twinhelix.org That is one option. The new line Digium cards are on par with Sangoma as far as IRQ issues. I really like Sangoma's lifetime warranty though. I don't think Digium has countered that bold move. I would try the RMA and if that doesn't work, you can always pickup a decent last year's model server at http://www.surpluscomputers.com/featured-hardware/cg-69/servers.html For a basic asterisk server or PBX with nothing special going on, any of these servers are more than enough, even overkill. No affiliation, I have to say the shipping is high and they are slow to ship but the prices are great, never had an issue with any of their boxen (dozens, knock on wood) I don't really know what you mean about the new line Digium cards.. which models are in this new line? the server i'm using is hardly new, it's one of the older DL145s; so i don't think this would help much! I've tried swapping the card in the slots. no help :( -- Tom O'Connor http://www.twinhelix.org t...@twinhelix.org Well I guess if I were you, I would stop posting woe is me to the list and call Digium. They do have support people just waiting for your call, you know? If they cannot help, then buy a better server. They are dirt cheap. Cheaper than the time you are wasting. I am not sure why you are opposed to taking suggestions and just replying with negatives. I expect your next reply to be *SOLVED* Echo and static on PRI with errors But somehow I doubt it. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- 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] UK Vodafone femtocells now available
For those of you who have been waiting for ATT to announce the public availability of their femtocell appliance in order to fix the shitty ATT network coverage this will interest you. Vodafone Access Gateway (femtocell) launched in UK http://www.abiresearch.com/Blog/Wireless_Blog/635 Its July 1st and Vodafone have officially launched their access gateway product in the UK. For those who are wondering what this is - its a femtocell with a curious title. Although Vodafone had made a press announcement at the Femto World Summit last week, its nice to see how they are packaging and presenting it to customers here in the UK. Some initial observations 1. Apart from buying it directly for £160, it is being bundled with var... Read More http://www.abiresearch.com/Blog/Wireless_Blog/635 I'm assuming it will be offered on similar commercial terms as I think if ATT charge a monthly fee people will riot - was hoping for a cheaper price than $US220 though. Regards, Dean Collins Cognation Inc d...@cognation.net mailto:d...@cognation.net +1-212-203-4357 New York +61-2-9016-5642 (Sydney in-dial). +44-20-3129-6001 (London 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
Re: [asterisk-users] Testing the manager.conf: sending and receiving commands
On Wed, 1 Jul 2009, bilal ghayyad wrote: Hi All; How can I test manager.conf? Can I telnet to the asterisk machine at the port 5038 and send and receive commands to test if the manager is working fine? How? Yes! RTFM would be a fine place to start - or at least the wiki: http://www.voip-info.org/wiki/view/Asterisk+manager+API I suggest preparing a file in one window with commands in it, then copy paste these into the telnet window while you're experimenting. 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] UK Vodafone femtocells now available
On Wed, 1 Jul 2009, Dean Collins wrote: For those of you who have been waiting for ATT to announce the public availability of their femtocell appliance in order to fix the shitty ATT network coverage this will interest you. It's getting a lot of press and a bit of a mixed reaction over here. Some are complaining that they shouldn't have to pay to extend the networks coverage, others wanting to jailbreak their iPhones to take advantage of poor O2 coverage where they are... (but moving to voda is a backward step IMO ;-) 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] UK Vodafone femtocells now available
agreed. extended o2 coverage would be very useful, especially for Wales! I like the idea, i don't like the idea of paying, if they want mobile traffic it should be possible to buy your own hardware controlled in the same method as wireless AP's allowing you to connect for free to the service and not be tied to a contract; or pay a very much reduced rate with an optional addon to your service for £2 or £3/month. Looking forward to seeing what the other networks will have to offer! 2009/7/1 Gordon Henderson gordon+aster...@drogon.netgordon%2baster...@drogon.net On Wed, 1 Jul 2009, Dean Collins wrote: For those of you who have been waiting for ATT to announce the public availability of their femtocell appliance in order to fix the shitty ATT network coverage this will interest you. It's getting a lot of press and a bit of a mixed reaction over here. Some are complaining that they shouldn't have to pay to extend the networks coverage, others wanting to jailbreak their iPhones to take advantage of poor O2 coverage where they are... (but moving to voda is a backward step IMO ;-) 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 ___ -- 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] UK Vodafone femtocells now available
2009/7/1 Geraint Lee gera...@gmail.com: agreed. extended o2 coverage would be very useful, especially for Wales! I like the idea, i don't like the idea of paying, if they want mobile traffic it should be possible to buy your own hardware controlled in the same method as wireless AP's allowing you to connect for free to the service and not be tied to a contract; or pay a very much reduced rate with an optional addon to your service for £2 or £3/month. I thought I read on the Vodafone site it was to be included with any 3g contracts over £25 per month? Maybe I misread? Mike Looking forward to seeing what the other networks will have to offer! 2009/7/1 Gordon Henderson gordon+aster...@drogon.net On Wed, 1 Jul 2009, Dean Collins wrote: For those of you who have been waiting for ATT to announce the public availability of their femtocell appliance in order to fix the shitty ATT network coverage this will interest you. It's getting a lot of press and a bit of a mixed reaction over here. Some are complaining that they shouldn't have to pay to extend the networks coverage, others wanting to jailbreak their iPhones to take advantage of poor O2 coverage where they are... (but moving to voda is a backward step IMO ;-) 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 ___ -- 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] Registrations problems to SIP-provider.
Hello List, I'm having problems with registrating my Asterisk-server to the SIP-provider. Yesterday all worked fine, this evening I cannot call out. What can be wrong ? This is my registration in sip.conf : register = 092779077:x...@85.119.188.3 This the output of SIP show peers : asterisk*CLI sip show peers Name/username HostDyn Nat ACL Port Status twinkle-candy/twinkle-can (Unspecified)D 0 UNKNOWN twinkle-jonas/twinkle-jon (Unspecified)D 0 UNKNOWN grandstream/grandstream192.168.1.13 D 5060 OK (35 ms) 3starsnet/09277907785.119.188.3 N 5060 UNREACHABLE This is the output of SIP debug : --- [Jul 1 21:08:37] NOTICE[15920]: chan_sip.c:7683 sip_reg_timeout:-- Registration for '092779...@85.119.188.3' timed out, trying again (Attempt #2) [Jul 1 21:08:37] REGISTER 12 headers, 0 lines [Jul 1 21:08:37] Reliably Transmitting (no NAT) to 85.119.188.3:5060: REGISTER sip:85.119.188.3 SIP/2.0 Via: SIP/2.0/UDP 78.22.164.52:5060;branch=z9hG4bK0fa45d19;rport From: sip:092779...@85.119.188.3;tag=as3306590c To: sip:092779...@85.119.188.3 Call-ID: 5d983c167b08b76b6211954c63c2a...@127.0.0.1 CSeq: 104 REGISTER User-Agent: Asterisk-jocan Max-Forwards: 70 Expires: 120 Contact: sip:s...@78.22.164.52 Event: registration Content-Length: 0 --- [Jul 1 21:08:37] Really destroying SIP dialog '5d983c167b08b76b6211954c63c2a...@127.0.0.1' Method: REGISTER [Jul 1 21:08:38] Retransmitting #1 (no NAT) to 85.119.188.3:5060: REGISTER sip:85.119.188.3 SIP/2.0 Via: SIP/2.0/UDP 78.22.164.52:5060;branch=z9hG4bK0fa45d19;rport From: sip:092779...@85.119.188.3;tag=as3306590c To: sip:092779...@85.119.188.3 Call-ID: 5d983c167b08b76b6211954c63c2a...@127.0.0.1 CSeq: 104 REGISTER User-Agent: Asterisk-jocan Max-Forwards: 70 Expires: 120 Contact: sip:s...@78.22.164.52 Event: registration Content-Length: 0 --- [Jul 1 21:08:39] Retransmitting #2 (no NAT) to 85.119.188.3:5060: REGISTER sip:85.119.188.3 SIP/2.0 Via: SIP/2.0/UDP 78.22.164.52:5060;branch=z9hG4bK0fa45d19;rport From: sip:092779...@85.119.188.3;tag=as3306590c To: sip:092779...@85.119.188.3 Call-ID: 5d983c167b08b76b6211954c63c2a...@127.0.0.1 CSeq: 104 REGISTER User-Agent: Asterisk-jocan Max-Forwards: 70 Expires: 120 Contact: sip:s...@78.22.164.52 Event: registration Content-Length: 0 --- [Jul 1 21:08:39] Reliably Transmitting (NAT) to 85.119.188.3:5060: OPTIONS sip:85.119.188.3 SIP/2.0 Via: SIP/2.0/UDP 78.22.164.52:5060;branch=z9hG4bK671f78b3;rport From: asterisk sip:aster...@78.22.164.52;tag=as6e1c81b5 To: sip:85.119.188.3 Contact: sip:aster...@78.22.164.52 Call-ID: 070856d002110e3b6422a38747e54...@78.22.164.52 CSeq: 102 OPTIONS User-Agent: Asterisk-jocan Max-Forwards: 70 Date: Wed, 01 Jul 2009 19:08:39 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Jul 1 21:08:40] Retransmitting #1 (NAT) to 85.119.188.3:5060: OPTIONS sip:85.119.188.3 SIP/2.0 Via: SIP/2.0/UDP 78.22.164.52:5060;branch=z9hG4bK671f78b3;rport From: asterisk sip:aster...@78.22.164.52;tag=as6e1c81b5 To: sip:85.119.188.3 Contact: sip:aster...@78.22.164.52 Call-ID: 070856d002110e3b6422a38747e54...@78.22.164.52 CSeq: 102 OPTIONS User-Agent: Asterisk-jocan Max-Forwards: 70 Date: Wed, 01 Jul 2009 19:08:39 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Jul 1 21:08:41] Retransmitting #3 (no NAT) to 85.119.188.3:5060: REGISTER sip:85.119.188.3 SIP/2.0 Via: SIP/2.0/UDP 78.22.164.52:5060;branch=z9hG4bK0fa45d19;rport From: sip:092779...@85.119.188.3;tag=as3306590c To: sip:092779...@85.119.188.3 Call-ID: 5d983c167b08b76b6211954c63c2a...@127.0.0.1 CSeq: 104 REGISTER User-Agent: Asterisk-jocan Max-Forwards: 70 Expires: 120 Contact: sip:s...@78.22.164.52 Event: registration Content-Length: 0 --- [Jul 1 21:08:41] Retransmitting #2 (NAT) to 85.119.188.3:5060: OPTIONS sip:85.119.188.3 SIP/2.0 Via: SIP/2.0/UDP 78.22.164.52:5060;branch=z9hG4bK671f78b3;rport From: asterisk sip:aster...@78.22.164.52;tag=as6e1c81b5 To: sip:85.119.188.3 Contact: sip:aster...@78.22.164.52 Call-ID: 070856d002110e3b6422a38747e54...@78.22.164.52 CSeq: 102 OPTIONS User-Agent: Asterisk-jocan Max-Forwards: 70 Date: Wed, 01 Jul 2009 19:08:39 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Jul 1 21:08:42] Retransmitting #3 (NAT) to 85.119.188.3:5060: OPTIONS sip:85.119.188.3 SIP/2.0 Via: SIP/2.0/UDP 78.22.164.52:5060;branch=z9hG4bK671f78b3;rport From: asterisk sip:aster...@78.22.164.52;tag=as6e1c81b5 To: sip:85.119.188.3 Contact: sip:aster...@78.22.164.52 Call-ID: 070856d002110e3b6422a38747e54...@78.22.164.52 CSeq: 102 OPTIONS User-Agent: Asterisk-jocan Max-Forwards: 70 Date: Wed, 01 Jul 2009 19:08:39 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length:
Re: [asterisk-users] * as VM for legacy PBX?
On Wed, 2009-07-01 at 13:05 -0400, Ken D'Ambrosio wrote: Sounds like good stuff, but my most substantial concerns involved things like MWI: is asterisk able to push that back to the PBX? Does your existing PBX use SMDI to interface with your current voicemail system? If so, recent versions of Asterisk (1.6.0 and later, if I recall) support SMDI. -- Jared Smith Training Manager Digium, Inc. ___ -- 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] Testing the manager.conf: sending and receiving commands
On Wed, 2009-07-01 at 10:25 -0700, bilal ghayyad wrote: Can I telnet to the asterisk machine at the port 5038 and send and receive commands to test if the manager is working fine? Absolutely! How? 1) Make sure manager is enabled in manager.conf (enabled=yes in [general] section) 2) Create a manager user, and give that user permissions (see the sample section in manager.conf named [mark]) 3) Type manager reload from the Asterisk CLI 4) Telnet to port 5038, as shown below: [jsm...@mybox ~]$ telnet localhost 5038 Trying 127.0.0.1... Connected to localhost. Escape character is '^]'. Asterisk Call Manager/1.1 Action: Login Username: jsmith Secret: doughnuts Events: on ActionID: 12345 Response: Success ActionID: 12345 Message: Authentication accepted Action: ExtensionState Exten: 555 Context: lab ActionID: 987654321 Response: Success ActionID: 987654321 Message: Extension Status Exten: 555 Context: lab Hint: SIP/linksys Status: 0 -- Jared Smith Training Manager Digium, Inc. ___ -- 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] Registrations problems to SIP-provider.
jonas kellens wrote: Hello List, I'm having problems with registrating my Asterisk-server to the SIP-provider. Yesterday all worked fine, this evening I cannot call out. What can be wrong ? This is my registration in sip.conf : register = 092779077:x...@85.119.188.3 mailto:df6...@85.119.188.3 This the output of SIP show peers : asterisk*CLI sip show peers Name/username HostDyn Nat ACL Port Status twinkle-candy/twinkle-can (Unspecified)D 0 UNKNOWN twinkle-jonas/twinkle-jon (Unspecified)D 0 UNKNOWN grandstream/grandstream192.168.1.13 D 5060 OK (35 ms) 3starsnet/09277907785.119.188.3 N 5060 UNREACHABLE I'd say that your server is no longer able to access the SIP-provider. Confirm that you have network access first, then verify that you didn't make any changes to your configuration. If everything is good there, I would work with your provider to make sure the settings are correct. -- Dean Hoover Network Administrator Centurion, Inc. 262-317-5622 Phone dhoo...@centonline.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] Registrations problems to SIP-provider.
Reliably Transmitting (no NAT) and you are natted I presume ( Port 5060 is forwarded to the internal IP-address of my Asterisk-server). Another Belgian user :) Olivier jonas kellens a écrit : Hello List, I'm having problems with registrating my Asterisk-server to the SIP-provider. Yesterday all worked fine, this evening I cannot call out. What can be wrong ? This is my registration in sip.conf : register = 092779077:x...@85.119.188.3 This the output of SIP show peers : asterisk*CLI sip show peers Name/username Host Dyn Nat ACL Port Status twinkle-candy/twinkle-can (Unspecified) D 0 UNKNOWN twinkle-jonas/twinkle-jon (Unspecified) D 0 UNKNOWN grandstream/grandstream 192.168.1.13 D 5060 OK (35 ms) 3starsnet/092779077 85.119.188.3 N 5060 UNREACHABLE This is the output of SIP debug : --- [Jul 1 21:08:37] NOTICE[15920]: chan_sip.c:7683 sip_reg_timeout: -- Registration for '092779...@85.119.188.3' timed out, trying again (Attempt #2) [Jul 1 21:08:37] REGISTER 12 headers, 0 lines [Jul 1 21:08:37] Reliably Transmitting (no NAT) to 85.119.188.3:5060: REGISTER sip:85.119.188.3 SIP/2.0 Via: SIP/2.0/UDP 78.22.164.52:5060;branch=z9hG4bK0fa45d19;rport From: sip:092779...@85.119.188.3;tag=as3306590c To: sip:092779...@85.119.188.3 Call-ID: 5d983c167b08b76b6211954c63c2a...@127.0.0.1 CSeq: 104 REGISTER User-Agent: Asterisk-jocan Max-Forwards: 70 Expires: 120 Contact: sip:s...@78.22.164.52 Event: registration Content-Length: 0 --- [Jul 1 21:08:37] Really destroying SIP dialog '5d983c167b08b76b6211954c63c2a...@127.0.0.1' Method: REGISTER [Jul 1 21:08:38] Retransmitting #1 (no NAT) to 85.119.188.3:5060: REGISTER sip:85.119.188.3 SIP/2.0 Via: SIP/2.0/UDP 78.22.164.52:5060;branch=z9hG4bK0fa45d19;rport From: sip:092779...@85.119.188.3;tag=as3306590c To: sip:092779...@85.119.188.3 Call-ID: 5d983c167b08b76b6211954c63c2a...@127.0.0.1 CSeq: 104 REGISTER User-Agent: Asterisk-jocan Max-Forwards: 70 Expires: 120 Contact: sip:s...@78.22.164.52 Event: registration Content-Length: 0 --- [Jul 1 21:08:39] Retransmitting #2 (no NAT) to 85.119.188.3:5060: REGISTER sip:85.119.188.3 SIP/2.0 Via: SIP/2.0/UDP 78.22.164.52:5060;branch=z9hG4bK0fa45d19;rport From: sip:092779...@85.119.188.3;tag=as3306590c To: sip:092779...@85.119.188.3 Call-ID: 5d983c167b08b76b6211954c63c2a...@127.0.0.1 CSeq: 104 REGISTER User-Agent: Asterisk-jocan Max-Forwards: 70 Expires: 120 Contact: sip:s...@78.22.164.52 Event: registration Content-Length: 0 --- [Jul 1 21:08:39] Reliably Transmitting (NAT) to 85.119.188.3:5060: OPTIONS sip:85.119.188.3 SIP/2.0 Via: SIP/2.0/UDP 78.22.164.52:5060;branch=z9hG4bK671f78b3;rport From: "asterisk" sip:aster...@78.22.164.52;tag=as6e1c81b5 To: sip:85.119.188.3 Contact: sip:aster...@78.22.164.52 Call-ID: 070856d002110e3b6422a38747e54...@78.22.164.52 CSeq: 102 OPTIONS User-Agent: Asterisk-jocan Max-Forwards: 70 Date: Wed, 01 Jul 2009 19:08:39 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Jul 1 21:08:40] Retransmitting #1 (NAT) to 85.119.188.3:5060: OPTIONS sip:85.119.188.3 SIP/2.0 Via: SIP/2.0/UDP 78.22.164.52:5060;branch=z9hG4bK671f78b3;rport From: "asterisk" sip:aster...@78.22.164.52;tag=as6e1c81b5 To: sip:85.119.188.3 Contact: sip:aster...@78.22.164.52 Call-ID: 070856d002110e3b6422a38747e54...@78.22.164.52 CSeq: 102 OPTIONS User-Agent: Asterisk-jocan Max-Forwards: 70 Date: Wed, 01 Jul 2009 19:08:39 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Jul 1 21:08:41] Retransmitting #3 (no NAT) to 85.119.188.3:5060: REGISTER sip:85.119.188.3 SIP/2.0 Via: SIP/2.0/UDP 78.22.164.52:5060;branch=z9hG4bK0fa45d19;rport From: sip:092779...@85.119.188.3;tag=as3306590c To: sip:092779...@85.119.188.3 Call-ID: 5d983c167b08b76b6211954c63c2a...@127.0.0.1 CSeq: 104 REGISTER User-Agent: Asterisk-jocan Max-Forwards: 70 Expires: 120 Contact: sip:s...@78.22.164.52 Event: registration Content-Length: 0 --- [Jul 1 21:08:41] Retransmitting #2 (NAT) to 85.119.188.3:5060: OPTIONS sip:85.119.188.3 SIP/2.0 Via: SIP/2.0/UDP 78.22.164.52:5060;branch=z9hG4bK671f78b3;rport From: "asterisk" sip:aster...@78.22.164.52;tag=as6e1c81b5 To: sip:85.119.188.3 Contact: sip:aster...@78.22.164.52 Call-ID: 070856d002110e3b6422a38747e54...@78.22.164.52 CSeq: 102 OPTIONS User-Agent: Asterisk-jocan Max-Forwards: 70 Date: Wed, 01 Jul 2009 19:08:39 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Jul 1 21:08:42]
[asterisk-users] g729a compatibility
Hello! I have a sip device that is sending in the SDP: rtpmap:98 g729a It does not seem like Asterisk is negotiating the codec properly, because while the call rings, the rtp lines fail. However, on other sip devices that have rtpmap:18 g729 in their SDP, things work fine with Digium's commercial g729 license. How do I get 98 g729a recognized by Asterisk? Thanks, Elliot ___ -- 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] * as VM for legacy PBX?
Wow. Thanks for all the replies! Something just occurred to me, though: which side would be FXO, and which side would be FXS? The PBX? Or the Asterisk/VM side? Thanks again for all the info! -Ken On Wed, July 1, 2009 3:36 pm, Jared Smith wrote: On Wed, 2009-07-01 at 13:05 -0400, Ken D'Ambrosio wrote: Sounds like good stuff, but my most substantial concerns involved things like MWI: is asterisk able to push that back to the PBX? Does your existing PBX use SMDI to interface with your current voicemail system? If so, recent versions of Asterisk (1.6.0 and later, if I recall) support SMDI. -- Jared Smith Training Manager Digium, Inc. ___ -- 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 message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ -- 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] g729a compatibility
Elliot Murdock wrote: Hello! I have a sip device that is sending in the SDP: rtpmap:98 g729a It does not seem like Asterisk is negotiating the codec properly, because while the call rings, the rtp lines fail. However, on other sip devices that have rtpmap:18 g729 in their SDP, things work fine with Digium's commercial g729 license. How do I get 98 g729a recognized by Asterisk? You don't. That's not a standards-compliant way of reporting G.729A in SDP. The RFC says it should be 'G729', but Asterisk also accepts 'G.729' and 'G729A'. It does not accept any lowercase form of the codec name. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- 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] * as VM for legacy PBX?
The PBX would be FXS since it originates the calls, * would be FXO since it only receives calls in this case. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ken D'Ambrosio Sent: Wednesday, July 01, 2009 4:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] * as VM for legacy PBX? Wow. Thanks for all the replies! Something just occurred to me, though: which side would be FXO, and which side would be FXS? The PBX? Or the Asterisk/VM side? Thanks again for all the info! -Ken On Wed, July 1, 2009 3:36 pm, Jared Smith wrote: On Wed, 2009-07-01 at 13:05 -0400, Ken D'Ambrosio wrote: Sounds like good stuff, but my most substantial concerns involved things like MWI: is asterisk able to push that back to the PBX? Does your existing PBX use SMDI to interface with your current voicemail system? If so, recent versions of Asterisk (1.6.0 and later, if I recall) support SMDI. -- Jared Smith Training Manager Digium, Inc. ___ -- 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 message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ -- 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] * as VM for legacy PBX?
2009/7/1 Ken D'Ambrosio k...@jots.org Wow. Thanks for all the replies! Something just occurred to me, though: which side would be FXO, and which side would be FXS? The PBX? Or the Asterisk/VM side? It seems PBX should be equiped with FXO interface(s) and Asterisk with FXS ones. Thanks again for all the info! -Ken ers 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] Extension status as XML for an Aastra 57i
2009/7/1 Jonathan Moore supermegat...@gmail.com On Wed, Jul 1, 2009 at 1:10 AM, Olivieroza-4...@myamail.com wrote: The 57i phone has 6 soft buttons which can show the status of at least 16 phones (if you do not want to use the rest of the soft buttons which would give you another 16). Are you sure of that ? How can you set more than one single phone to light on or off a given BLF ? With a single button, I agree you can query more than one phone status but the associated light can't display more than one phone or am I missing something ? On the 57i, there are 6 soft buttons above and the screen, and 6 below. The top set can have up to 10 configurations, when you add more than 6, the bottom right button changes to Next.. and scrolls the screen over. The bottom can have up to 20, with the same next button. Each of these keys can be configured as BLF keys. True but how can a single light be blinking because extension 1001 is receiving a call and at the same time, be turned on because extension 1002 is on call ? Maybe typing on Next button would alternatively show extension 1001 or 1002 status, but without a press on this Next key, a user can't be aware of all status changes as he would if equiped with dedicated BLF. But of course, this might be enough for some (operators, ...) but not for all (group secretary ...). -jonathan ___ -- 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] * as VM for legacy PBX?
2009/7/1 Danny Nicholas da...@debsinc.com The PBX would be FXS since it originates the calls, * would be FXO since it only receives calls in this case. Yes you're right : if Asterisk behaves like a phone, it should plus into PBX's FXS ports (and so be equiped with FXO ports). Sorry, for my previous misleading answer and thanks for correcting it ... -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ken D'Ambrosio Sent: Wednesday, July 01, 2009 4:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] * as VM for legacy PBX? Wow. Thanks for all the replies! Something just occurred to me, though: which side would be FXO, and which side would be FXS? The PBX? Or the Asterisk/VM side? Thanks again for all the info! -Ken On Wed, July 1, 2009 3:36 pm, Jared Smith wrote: On Wed, 2009-07-01 at 13:05 -0400, Ken D'Ambrosio wrote: Sounds like good stuff, but my most substantial concerns involved things like MWI: is asterisk able to push that back to the PBX? Does your existing PBX use SMDI to interface with your current voicemail system? If so, recent versions of Asterisk (1.6.0 and later, if I recall) support SMDI. -- Jared Smith Training Manager Digium, Inc. ___ -- 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 message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ -- 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] * as VM for legacy PBX?
Looks like you might be getting conflicting information. The important thing is the * ports must be opposite what the PBX ports are. Odds are, the ports on the PBX used for voice-mail are extension ports--they look like central office lines (PSTN POTS), providing dial tone. The ports on the voice mail server look like stations (telephones). In * lingo, an FXO (Foreign eXchange Office) port CONNECTS TO an Office port, so it LOOKS LIKE a station. an FXS (Foreign eXchange Station) port CONNECTS TO a Station,, so it LOOKS LIKE a phone line. When you do the test suggested earlier, plugging a phone into a port on the PBX used for voice mail, you'll most likely get dial tone. When you attempt to reach someone's voice mail through the PBX, the phone will ring. This confirms that the PBX is acting like an Office, so you want * to act like a Station. You would install FXO ports in the * system. --Don Don Kelly PCF Corp People Come First 651 842-1000 888 Don Kell(y) 651 842-1001 fax -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ken D'Ambrosio Sent: Wednesday, July 01, 2009 4:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] * as VM for legacy PBX? Wow. Thanks for all the replies! Something just occurred to me, though: which side would be FXO, and which side would be FXS? The PBX? Or the Asterisk/VM side? Thanks again for all the info! -Ken On Wed, July 1, 2009 3:36 pm, Jared Smith wrote: On Wed, 2009-07-01 at 13:05 -0400, Ken D'Ambrosio wrote: Sounds like good stuff, but my most substantial concerns involved things like MWI: is asterisk able to push that back to the PBX? Does your existing PBX use SMDI to interface with your current voicemail system? If so, recent versions of Asterisk (1.6.0 and later, if I recall) support SMDI. -- Jared Smith Training Manager Digium, Inc. ___ -- 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 message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ -- 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] * as VM for legacy PBX?
On Wed, Jul 1, 2009 at 3:36 PM, Jared Smithjsm...@digium.com wrote:
On Wed, 2009-07-01 at 13:05 -0400, Ken D'Ambrosio wrote:
Sounds like good stuff, but my most substantial concerns involved things
like MWI: is asterisk able to push that back to the PBX?
Does your existing PBX use SMDI to interface with your current voicemail
system? If so, recent versions of Asterisk (1.6.0 and later, if I
recall) support SMDI.
--
Jared Smith
Training Manager
Digium, Inc.
Here is a pretty classic integration that works with most systems that
just send DTMF.
http://www.voip-info.org/wiki/view/Avaya+or+Lucent+Magix+Voicemail+Integration
Is a pretty good how to all the way to MWI. It didn't work exactly
like the wiki said though. To catch the DTMF being sent by the PBX I
used
Answer()
WaitExten(1)
NoOp(${WaitExten})
With the PBX plugged into asterisk I would place a call and on the
console, I could see what digits were being sent with the NoOp line
and adjust accordingly.
Then this piece of genius just needed a slight bit of tweaking
http://mikecathey.com/code/vmnotify/?M=A
Thanks,
Steve Totaro
--
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)
___
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[asterisk-users] /var/lib/asterisk/sounds does not exist
Hi All; I download asterisk, compiled it and install it, but not finding the sounds file (/var/lib/asterisk/sounds), what could be the reason and how I can have it without repeating every thing? My asterisk version is: Asterisk 1.4.25 Regards Bilal ___ -- 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] IAX2 help needed...
On Tue, Jun 30, 2009 at 10:47 AM, Ade Vickersaster...@solutionengineers.com wrote: I run a phone in a remote office using the IAX2 protocol. It mostly works fine; except that every 5 mins it loses connection with Asterisk, before reconnecting 30 seconds later; rinse repeat. I used to have that happen a lot. I had no idea what caused it, or what the solution was. I ended up using SIP instead. Problem no longer existed, but I never found a solution. ___ -- 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 1.6 WaitForSilence Problem
On Tue, Jun 30, 2009 at 9:21 AM, Deric Pagederic.p...@nisc.coop wrote: I've set up an outbound .call system for customer callbacks and the like. Calls are going out over analog lines and I'm trying to use the WaitForSilence routine to make sure the phone has stopped ringing before starting message playback. The problem is that if I set the first argument of WaitForSilence to anything other than 1, WaitForSilence never exits. I would suggest that WaitForSilence isn't a very good tool to 'make sure the phone has stopped ringing', especially since it seems like what you really want is a smart way to detect whether you have a human on the phone. You may prefer Answering Machine Detection, known in dialplans as AMD(). If you honestly just want a wait, put in a wait. If somebody starts talking by saying hello, WaitForSilence() may not be what you want. Have you recorded some tests and listened back to them? ___ -- 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] IAX2 help needed...
Check out http://www.voip-info.org/wiki/view/Asterisk+iax+qualify. I've ran into problems with home routers not keeping the connection alive, udp timeouts most likely. These options particularly, the qualifyfreqnotok will have asterisk send out a poke to the soft phone if it reports the phone is offline. Might not be the best for a soft phone which is not always in use, but we use it on our iax trunks. From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of David Backeberg [dbackeb...@gmail.com] Sent: Wednesday, July 01, 2009 9:02 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] IAX2 help needed... On Tue, Jun 30, 2009 at 10:47 AM, Ade Vickersaster...@solutionengineers.com wrote: I run a phone in a remote office using the IAX2 protocol. It mostly works fine; except that every 5 mins it loses connection with Asterisk, before reconnecting 30 seconds later; rinse repeat. I used to have that happen a lot. I had no idea what caused it, or what the solution was. I ended up using SIP instead. Problem no longer existed, but I never found a solution. ___ -- 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 email and its attachments may be confidential and are intended solely for the use of the individual or parties' to whom it is addressed. All comments are solely those of the author and do not necessarily represent those of Ignition. If you are not the intended recipient of this email and its attachments, you must take no action based upon them, nor must you copy or show them to anyone. Please contact the sender if you believe you have received this email in error. Thanks for considering the environmental impact before printing this email. ___ -- 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] Extension status as XML for an Aastra 57i
On Wed, Jul 1, 2009 at 4:40 PM, Olivieroza-4...@myamail.com wrote: True but how can a single light be blinking because extension 1001 is receiving a call and at the same time, be turned on because extension 1002 is on call ? Maybe typing on Next button would alternatively show extension 1001 or 1002 status, but without a press on this Next key, a user can't be aware of all status changes as he would if equiped with dedicated BLF. That's the problem with using the buttons in this manner. Exactly as you said, the light will only represent a single item at any one time. To see more than one you have to switch what the light is representing at that moment. Of course, as you point out this may not work out. If that's the case, your next option is to go with the 560m or 536m depending. Of course, you can also add several of to each 57i phone (up to 3, IIRC). But of course, this might be enough for some (operators, ...) but not for all (group secretary ...). No disagreement. It all comes down to how much you're willing to pay for the convenience. If you (or the users) don't want to switch screens, you get a single, or more than one, sidecar. -jonathan ___ -- 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
