[asterisk-users] RTP timestamp modification during SIP video call
Hi, I'm using asterisk 1.14.19 I'm making a video call between two SIP end-points, using h263p and iLBC. I notice the video is jumpy and I believe the cause is due to RTP timestamps. The sending device is working at 8fps and correctly increases the timestamp by 11250 every frame. It appears that Asterisk is modifying the timestamps and generating new ones which sometimes increase by 11250, but sometimes have much longer delays. Should asterisk use the senders RTP timestamp? Maybe someone can explain why this is happening? Thanks, Dan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question about Dialing DTMF
Venefax wrote: I need to dial a DTMF string with the Dial function using the D(“DTMF”) function. What is the character for a delay? I mean, normally in other technologies we use the comma to mean “wait 200 ms “. Is there an equivalent in Asterisk? If it is the comma indeed, how many ms will the system wait for each comma? w -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question about Dialing DTMF
Alex Balashov wrote: Venefax wrote: I need to dial a DTMF string with the Dial function using the D(“DTMF”) function. What is the character for a delay? I mean, normally in other technologies we use the comma to mean “wait 200 ms “. Is there an equivalent in Asterisk? If it is the comma indeed, how many ms will the system wait for each comma? w The delay is 500 ms. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Semi-OT Satellite?
Alex Balashov [EMAIL PROTECTED] writes: Yes, indeed. Encapsulation protocols such as IPSec/GRE won't work at all over high RTT latency (= 400 ms). Sure they will. They just won't benefit from TCP acceleration performed by the satellite company. Sadly TCP itself is very slow with high RTT, so without TCP acceleration the satellite link is a bit useless. I don't see why it should be a problem for VoIP though, there's no VoIP acceleration technology. /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Realtime SIP
Probably I did not read well the information I am concerning, if I am going to use ARA for the SIP and I have register = user:secret:[EMAIL PROTECTED]:port/extension how I should input that line? If I am going to delete it from the DB I am forced to reload everything or there is a way to tell asterisk to remove only a particular entry? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dial Plan Help
I'd like to do the following can someone guide me on how to accomplish this?
Call comes in via PRI and tries to go out via SIP if for some reason the ISP
is down and the call can not go out i want it to fail over and send the same
call through a different PRI.
I was thinking something like this:
exten=_X.,1,Dial(SIP/[EMAIL PROTECTED])
exten=_X.,2,Dial,Zap/g2/${EXTEN}; I only want it to go here if it
was unable to send the call via SIP (if the first priority failed), but if
it did go through sip then it should just hangup the call when the person is
done speaking.
Thanks,
Jon
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Re: [asterisk-users] Dial Plan Help
On Sun, Aug 24, 2008 at 8:11 AM, Jon Weisman [EMAIL PROTECTED] wrote:
I'd like to do the following can someone guide me on how to accomplish this?
Call comes in via PRI and tries to go out via SIP if for some reason the ISP
is down and the call can not go out i want it to fail over and send the same
call through a different PRI.
I was thinking something like this:
exten=_X.,1,Dial(SIP/[EMAIL PROTECTED])
exten=_X.,2,Dial,Zap/g2/${EXTEN}; I only want it to go here if it
was unable to send the call via SIP (if the first priority failed), but if
it did go through sip then it should just hangup the call when the person is
done speaking.
Thanks,
Jon
Jon,
This should work just fine with the correct dial syntax, after a call
ends, the exten goes to h for hangup rather then progressing further
down the priority for the initial exten.
Double check your syntax.
http://www.voip-info.org/wiki-Asterisk+cmd+Dial
Thanks,
Steve Totaro
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Re: [asterisk-users] Dial Plan Help
John,
This is the default behaviour anyway. If Dial() is successful,
execution of subsequent priorities in the dial plan for that extension
is not resumed. It'll only fall through to the other priorities if
Dial() fails.
I do, however, suggest supplying a timeout argument to your Dial()s.
-- Alex
Jon Weisman wrote:
I'd like to do the following can someone guide me on how to accomplish this?
Call comes in via PRI and tries to go out via SIP if for some reason the ISP
is down and the call can not go out i want it to fail over and send the same
call through a different PRI.
I was thinking something like this:
exten=_X.,1,Dial(SIP/[EMAIL PROTECTED])
exten=_X.,2,Dial,Zap/g2/${EXTEN}; I only want it to go here if it
was unable to send the call via SIP (if the first priority failed), but if
it did go through sip then it should just hangup the call when the person is
done speaking.
--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599
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[asterisk-users] entering a password to have access to a sip account?!
Hi all,
i;m obviously a newbie, its been 2 days that im trying to figure out a way to
deny a specific extension (300) from calling another specific extensions (03)
except if the caller punch a specified password.. sorry if im not explaining
myself well.. heres an example:
i called my pstn line(with 300 as its sip account), an attendant answers and
asks me to punch in an extension number right now if i dial 03 it rings at
the other end! though i dont want that to happen! i want to set asterisk up in
a way tht if i dial 03 from 300 to ask me for a password... or it wont let
the line go through!
can anyone guide me through this issue! im really going crazy to get this done!
any help would truly and utterly be appreciated:)
ps: find below my extensions.conf
[sipura-line]
exten = 301,1,Answer() ; Answer inbound calls
exten = 301,2,Playback(silence/1)
exten = 301,3,Background(simzy1) ; input an extension
exten = 301,4,WaitExten(8)
exten = 301,5,Dial(SIP/100,15) ; goes to operator
exten = 301,4,Wait(8)
include = spa
exten = _XXX,6,VoiceMail([EMAIL PROTECTED])
exten = 301,n,Hangup()
[spa]
exten =_301,1,GoTo(sipura-line,${EXTEN},1)
exten = _1XX,1,Dial(SIP/${EXTEN},20) ;each ring equals to 5 seconds so it will
ring 3 times
exten = _1XX,2,VoiceMail([EMAIL PROTECTED]) ; direct 2 voicemail box if line
is busy or unavailable
exten = _1XX,3,HangUp()
exten = _2XX,1,Dial(SIP/${EXTEN},20) ;each ring equals to 5 seconds so it will
ring 3 times
exten = _2XX,2,VoiceMail([EMAIL PROTECTED]) ; directs to voicemail box if line
is busy or unavailable
exten = _2XX,3,HangUp()
exten = _3XX,1,Dial(SIP/${EXTEN},20) ; each ring equals to 5 seconds so it
will ring 3 times
exten = _3XX,2,VoiceMail([EMAIL PROTECTED]) ; directs 2 voicemail box if line
is busy or unavailable
exten = _3XX,3,HangUp()
exten =_01,1,Dial(SIP/$(EXTEN)@300) ; old ogero line
;exten =_01,2,Set(TIMEOUT(absolute)=5)
exten =_02,1,Dial(SIP/$(EXTEN)@304) ; new ogero line
exten =_03,1,Dial(SIP/$(EXTEN)@305) ; samer
exten =_04,1,Dial(SIP/$(EXTEN)@306) ; gilberte
exten =_05,1,Dial(SIP/$(EXTEN)@307) ; conference
exten =_06,1,Dial(SIP/$(EXTEN)@308) ; line 4
exten = 303,1,VoicemailMain ; voicemail box to be redirected to
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Re: [asterisk-users] entering a password to have access to a sip account?!
You want to use Authenticate() between answer and dial.
http://www.google.com/search?q=asterisk+authenticateie=utf-8oe=utf-8aq=trls=org.mozilla:en-US:officialclient=firefox-a
Thanks,
Steve Totaro
On Sun, Aug 24, 2008 at 11:26 AM, RoLaNd RoLaNd [EMAIL PROTECTED] wrote:
Hi all,
i;m obviously a newbie, its been 2 days that im trying to figure out a way
to deny a specific extension (300) from calling another specific extensions
(03) except if the caller punch a specified password.. sorry if im not
explaining myself well.. heres an example:
i called my pstn line(with 300 as its sip account), an attendant answers and
asks me to punch in an extension number right now if i dial 03 it rings at
the other end! though i dont want that to happen! i want to set asterisk up
in a way tht if i dial 03 from 300 to ask me for a password... or it
wont let the line go through!
can anyone guide me through this issue! im really going crazy to get this
done! any help would truly and utterly be appreciated:)
ps: find below my extensions.conf
[sipura-line]
exten = 301,1,Answer() ; Answer inbound calls
exten = 301,2,Playback(silence/1)
exten = 301,3,Background(simzy1) ; input an extension
exten = 301,4,WaitExten(8)
exten = 301,5,Dial(SIP/100,15) ; goes to operator
exten = 301,4,Wait(8)
include = spa
exten = _XXX,6,VoiceMail([EMAIL PROTECTED])
exten = 301,n,Hangup()
[spa]
exten =_301,1,GoTo(sipura-line,${EXTEN},1)
exten = _1XX,1,Dial(SIP/${EXTEN},20) ;each ring equals to 5 seconds so it
will ring 3 times
exten = _1XX,2,VoiceMail([EMAIL PROTECTED]) ; direct 2 voicemail box if line
is busy or unavailable
exten = _1XX,3,HangUp()
exten = _2XX,1,Dial(SIP/${EXTEN},20) ;each ring equals to 5 seconds so it
will ring 3 times
exten = _2XX,2,VoiceMail([EMAIL PROTECTED]) ; directs to voicemail box if
line is busy or unavailable
exten = _2XX,3,HangUp()
exten = _3XX,1,Dial(SIP/${EXTEN},20) ; each ring equals to 5 seconds so it
will ring 3 times
exten = _3XX,2,VoiceMail([EMAIL PROTECTED]) ; directs 2 voicemail box if
line is busy or unavailable
exten = _3XX,3,HangUp()
exten =_01,1,Dial(SIP/$(EXTEN)@300) ; old ogero line
;exten =_01,2,Set(TIMEOUT(absolute)=5)
exten =_02,1,Dial(SIP/$(EXTEN)@304) ; new ogero line
exten =_03,1,Dial(SIP/$(EXTEN)@305) ; samer
exten =_04,1,Dial(SIP/$(EXTEN)@306) ; gilberte
exten =_05,1,Dial(SIP/$(EXTEN)@307) ; conference
exten =_06,1,Dial(SIP/$(EXTEN)@308) ; line 4
exten = 303,1,VoicemailMain ; voicemail box to be redirected to
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[asterisk-users] Asterisk - forcing PSTN Line ON Hook
Is there a way to force PSTN line on ATA (Linksys / Sipura) to go ON hood via dial plan? I have two Linksys units one connected to old POTS line and one connected via Shaw Digital Phone. The one connected to Shaw Digital Phone the PSTN Line fails to go ON Hook at the end of the conversation. I know the caller (PSTN caller or VoIP caller), if the call is not a ring-thru-Line-1 or forwarded call can enter **# to force the SPA to take the FXO port on-hook to disconnect the PSTN line. But I'm the receiver of the call. The problem is that if the PSTN line fails to disconnect the call goes to Voice mail. Standard POTS line works perfectly the PSTN Line goes ON Hook every time at the end of the call but not the Shaw Digital Phone -- #Joseph GPG KeyID: ED0E1FB7 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] entering a password to have access to a sip account?!
Hello Roland,
You can use the cmd Read for this.
http://www.voip-info.org/wiki/view/Asterisk+cmd+Read
Pretty straight forward. Whenever you need to accept DTMF input from the user
collect the required digits using Read; check the collected digits; if yes jump
to required extension; else reject user or whatever you want to do.
I could've written out the dialplan, but well... you are a newbie you said, so
you gotta learn ;-) .
Hope this helps.
- Ben.
--- On Sun, 8/24/08, RoLaNd RoLaNd [EMAIL PROTECTED] wrote:
From: RoLaNd RoLaNd [EMAIL PROTECTED]
Subject: [asterisk-users] entering a password to have access to a sip
account?!
To: asterisk-users@lists.digium.com
Date: Sunday, August 24, 2008, 3:26 PM
Hi all,
i;m obviously a newbie, its been 2 days that im trying to
figure out a way to deny a specific extension (300) from
calling another specific extensions (03) except if the
caller punch a specified password.. sorry if im not
explaining myself well.. heres an example:
i called my pstn line(with 300 as its sip account), an
attendant answers and asks me to punch in an extension
number right now if i dial 03 it rings at the
other end! though i dont want that to happen! i want to set
asterisk up in a way tht if i dial 03 from
300 to ask me for a password... or it wont let
the line go through!
can anyone guide me through this issue! im really going
crazy to get this done! any help would truly and utterly be
appreciated:)
ps: find below my extensions.conf
[sipura-line]
exten = 301,1,Answer() ; Answer inbound calls
exten = 301,2,Playback(silence/1)
exten = 301,3,Background(simzy1) ; input an extension
exten = 301,4,WaitExten(8)
exten = 301,5,Dial(SIP/100,15) ; goes to operator
exten = 301,4,Wait(8)
include = spa
exten = _XXX,6,VoiceMail([EMAIL PROTECTED])
exten = 301,n,Hangup()
[spa]
exten =_301,1,GoTo(sipura-line,${EXTEN},1)
exten = _1XX,1,Dial(SIP/${EXTEN},20) ;each ring equals
to 5 seconds so it will ring 3 times
exten = _1XX,2,VoiceMail([EMAIL PROTECTED]) ; direct 2
voicemail box if line is busy or unavailable
exten = _1XX,3,HangUp()
exten = _2XX,1,Dial(SIP/${EXTEN},20) ;each ring equals
to 5 seconds so it will ring 3 times
exten = _2XX,2,VoiceMail([EMAIL PROTECTED]) ; directs to
voicemail box if line is busy or unavailable
exten = _2XX,3,HangUp()
exten = _3XX,1,Dial(SIP/${EXTEN},20) ; each ring equals
to 5 seconds so it will ring 3 times
exten = _3XX,2,VoiceMail([EMAIL PROTECTED]) ; directs 2
voicemail box if line is busy or unavailable
exten = _3XX,3,HangUp()
exten =_01,1,Dial(SIP/$(EXTEN)@300) ; old ogero line
;exten =_01,2,Set(TIMEOUT(absolute)=5)
exten =_02,1,Dial(SIP/$(EXTEN)@304) ; new ogero line
exten =_03,1,Dial(SIP/$(EXTEN)@305) ; samer
exten =_04,1,Dial(SIP/$(EXTEN)@306) ; gilberte
exten =_05,1,Dial(SIP/$(EXTEN)@307) ; conference
exten =_06,1,Dial(SIP/$(EXTEN)@308) ; line 4
exten = 303,1,VoicemailMain ; voicemail box to be
redirected to
_
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[asterisk-users] MWI working perfectly. Shouldn't it be broken??
I have a Sipura 962 endpoint on Asterisk 1.4 where the MWI works perfectly, however my theory is that it should be broken. Obviously I'm wrong but Sip show subscriptions does not show the endpoint subscribing to the MWI status on Asterisk, even though all of the other endpoints on the system DO subscribe for their respective mailboxes, including SNOM Polycom endpoints. I'm confused. Isn't MWI subscription the method by which the device is setting its MWI? I know that Asterisk 1.6 is moving toward an event-driven model, but I'm running 1.4.21.1, so why is this working? I know that some smart cookie on this list will know the answer, but unfortunately I am not said cookie. FYI, it's not an issue of the subscription not YET subscribing etc. If I were to restart the system and endpoints, all the subs slowly show up one by one, but the 962 never does -- even after days, weeks, and months. Yet the MWI always works perfectly from the get-go. Thanks! -Karl ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [Xen-users] Xen 3.2.1 and PCI passthrough
Just tried upgrading to 3.3.0 by using the 3.2 spec and removing the patches etc, but after the system booted and it tried to start the images I got the message :- Error: Boot loader didn't return any data Has anybody else upgraded yet ? Regards, -- --[ UxBoD ]-- // PGP Key: curl -s http://www.splatnix.net/uxbod.asc | gpg --import // Fingerprint: F57A 0CBD DD19 79E9 1FCC A612 CB36 D89D 2C5A 3A84 // Keyserver: www.keyserver.net Key-ID: 0x2C5A3A84 // Phone: +44 845 869 2749 SIP Phone: [EMAIL PROTECTED] - --[ UxBoD ]-- [EMAIL PROTECTED] wrote: Yes I tried by creating a new initrd but it produced the same results :( Regards, -- 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - Which rackable case for mini-ITX boards ?
Gordon, I have decided to ditch using Xen for Asterisk and build a separate server. It will be used in the home office and this rack case looks great as I am going to get a small comms cabinet. We will be using a TDM400P to bring PSTN to the server, plus we have a few SNOM M3 phones around the house and in our offices. Would the CN1300 be capable for this ? and how much memory do you put in your servers ? Is that case capable of taking two SATA drives which I can then mirror ? or do you save any VMs etc to a separate server ? Regards, -- --[ UxBoD ]-- // PGP Key: curl -s http://www.splatnix.net/uxbod.asc | gpg --import // Fingerprint: F57A 0CBD DD19 79E9 1FCC A612 CB36 D89D 2C5A 3A84 // Keyserver: www.keyserver.net Key-ID: 0x2C5A3A84 // Phone: +44 845 869 2749 SIP Phone: [EMAIL PROTECTED] - Gordon Henderson [EMAIL PROTECTED] wrote: What processor do you need? ie. how many extensions/calls/PCI interface cards - and additional back-end stuff like MySQL, AGI, etc ? And what country are you in (although I'm guessing switzerland from those domains). I've built many with this combo: (from the UK, so you might be OK ordering from .ch, or find a local supplier) http://linitx.com/viewproduct.php?prodid=10902 + PSU + VIA CN1300 mobo + 1 PCI card... 230mm deep, so fits into small office comms cabinets rather than needing full racks, however I limit the CN1300's to 120 extensions - you might need more, or have other stuff like MySQL, etc. running so might want a beefier processor... (I don't run MySQL, or any other fancy AGI, etc. and I do build a custom system, kernel apps. targetted at the processor which runs entirely in RAM) I've also used this: http://linitx.com/viewproduct.php?prodid=10403 which can take 2 PCI cards via a riser, but I wasn't happy with it's performance - tried both Digium TDM400 ISDN30 cards at the same time, but the ISDN card kept on losing IRQs and due to time contraints, never had the time to get to the bottom of it. Gordon ___ -- 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MWI working perfectly. Shouldn't it be broken??
On Aug 24, 2008, at 11:29 AM, Karl Fife wrote: FYI, it's not an issue of the subscription not YET subscribing etc. If I were to restart the system and endpoints, all the subs slowly show up one by one, but the 962 never does -- even after days, weeks, and months. Yet the MWI always works perfectly from the get-go. Asterisk will send the NOTIFY for MWI even if the device doesn't subscribe, unless you tell it not to. This is necessary for some phones for MWI to work. If you _don't_ want Asterisk to do this, you can set the subscribemwi=yes option in sip.conf. This tells Asterisk to _only_ send MWI with an associated subscription. -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - Which rackable case for mini-ITX boards ?
Better solution for Comms Cabinet. http://www.hoffmanonline.com/product_catalog/product_detail.aspx?cat_1=34cat_2=2410cat_3=42359catID=42359itemID=3599 On Sun, Aug 24, 2008 at 12:38 PM, --[ UxBoD ]-- [EMAIL PROTECTED] wrote: Gordon, I have decided to ditch using Xen for Asterisk and build a separate server. It will be used in the home office and this rack case looks great as I am going to get a small comms cabinet. We will be using a TDM400P to bring PSTN to the server, plus we have a few SNOM M3 phones around the house and in our offices. Would the CN1300 be capable for this ? and how much memory do you put in your servers ? Is that case capable of taking two SATA drives which I can then mirror ? or do you save any VMs etc to a separate server ? Regards, -- --[ UxBoD ]-- // PGP Key: curl -s http://www.splatnix.net/uxbod.asc | gpg --import // Fingerprint: F57A 0CBD DD19 79E9 1FCC A612 CB36 D89D 2C5A 3A84 // Keyserver: www.keyserver.net Key-ID: 0x2C5A3A84 // Phone: +44 845 869 2749 SIP Phone: [EMAIL PROTECTED] - Gordon Henderson [EMAIL PROTECTED] wrote: What processor do you need? ie. how many extensions/calls/PCI interface cards - and additional back-end stuff like MySQL, AGI, etc ? And what country are you in (although I'm guessing switzerland from those domains). I've built many with this combo: (from the UK, so you might be OK ordering from .ch, or find a local supplier) http://linitx.com/viewproduct.php?prodid=10902 + PSU + VIA CN1300 mobo + 1 PCI card... 230mm deep, so fits into small office comms cabinets rather than needing full racks, however I limit the CN1300's to 120 extensions - you might need more, or have other stuff like MySQL, etc. running so might want a beefier processor... (I don't run MySQL, or any other fancy AGI, etc. and I do build a custom system, kernel apps. targetted at the processor which runs entirely in RAM) I've also used this: http://linitx.com/viewproduct.php?prodid=10403 which can take 2 PCI cards via a riser, but I wasn't happy with it's performance - tried both Digium TDM400 ISDN30 cards at the same time, but the ISDN card kept on losing IRQs and due to time contraints, never had the time to get to the bottom of it. Gordon ___ -- 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Andrew lathama Latham Principal TuxTone Inc. http://TuxTone.com [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - Which rackable case for mini-ITX boards ?
Just about anything bootable will serve you well enough. My home office is presently running of an HP T5700 thin client (1 GHz transmeta CPU, 256 MB RAM, 256 MB Flash DOM) running Astlinux. Cheap, diskless, fanless, silent, cool, reliable. Almost perfect IMHO. It's connected to a snom m3 system, 6 Polycom Aastra SIP phones, and one lowly Sipura SPA-2002. We don't have any POTS lines so no FXO card anymore. I did have a TDM400p at one point. Michael Graves On Sun, 24 Aug 2008 17:38:26 +0100 (BST), --[ UxBoD ]-- wrote: Gordon, I have decided to ditch using Xen for Asterisk and build a separate server. It will be used in the home office and this rack case looks great as I am going to get a small comms cabinet. We will be using a TDM400P to bring PSTN to the server, plus we have a few SNOM M3 phones around the house and in our offices. Would the CN1300 be capable for this ? and how much memory do you put in your servers ? Is that case capable of taking two SATA drives which I can then mirror ? or do you save any VMs etc to a separate server ? Regards, -- --[ UxBoD ]-- // PGP Key: curl -s http://www.splatnix.net/uxbod.asc | gpg --import // Fingerprint: F57A 0CBD DD19 79E9 1FCC A612 CB36 D89D 2C5A 3A84 // Keyserver: www.keyserver.net Key-ID: 0x2C5A3A84 // Phone: +44 845 869 2749 SIP Phone: [EMAIL PROTECTED] - Gordon Henderson [EMAIL PROTECTED] wrote: What processor do you need? ie. how many extensions/calls/PCI interface cards - and additional back-end stuff like MySQL, AGI, etc ? And what country are you in (although I'm guessing switzerland from those domains). I've built many with this combo: (from the UK, so you might be OK ordering from .ch, or find a local supplier) http://linitx.com/viewproduct.php?prodid=10902 + PSU + VIA CN1300 mobo + 1 PCI card... 230mm deep, so fits into small office comms cabinets rather than needing full racks, however I limit the CN1300's to 120 extensions - you might need more, or have other stuff like MySQL, etc. running so might want a beefier processor... (I don't run MySQL, or any other fancy AGI, etc. and I do build a custom system, kernel apps. targetted at the processor which runs entirely in RAM) I've also used this: http://linitx.com/viewproduct.php?prodid=10403 which can take 2 PCI cards via a riser, but I wasn't happy with it's performance - tried both Digium TDM400 ISDN30 cards at the same time, but the ISDN card kept on losing IRQs and due to time contraints, never had the time to get to the bottom of it. Gordon ___ -- 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Internal Virus Database is out of date. Checked by AVG - http://www.avg.com Version: 8.0.138 / Virus Database: 270.6.4/1617 - Release Date: 8/17/2008 12:58 PM -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - Which rackable case for mini-ITX boards ?
In the UK these look very nice :) going to get a dark grey/black mini one http://www.orionuk.biz/default.asp?productDetails=3 Regards, -- --[ UxBoD ]-- // PGP Key: curl -s http://www.splatnix.net/uxbod.asc | gpg --import // Fingerprint: F57A 0CBD DD19 79E9 1FCC A612 CB36 D89D 2C5A 3A84 // Keyserver: www.keyserver.net Key-ID: 0x2C5A3A84 // Phone: +44 845 869 2749 SIP Phone: [EMAIL PROTECTED] - Andrew Latham [EMAIL PROTECTED] wrote: Better solution for Comms Cabinet. http://www.hoffmanonline.com/product_catalog/product_detail.aspx?cat_1=34cat_2=2410cat_3=42359catID=42359itemID=3599 -- 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SECURITY QUESTION SANITY CHECK
SECURITY QUESTION SANITY CHECK: If only my SIP ports and a small range of RTP ports are facing the public internet, what is the method by which an evildoer would be able to do fraudulent long distance on my nickel? Would it REALLY be as simple as guessing the credentials for ANY of my local sip endpoints? Like most people, my local endpoint credentials would be easy to guess: Username is often just an extension number (101,2,3 etc), passwords are often found in the top 2000 common passwords list and more often in the few hundred thousand canonical words. I THINK the answer is YES, absolutely--Karl, go harden your installation! WHAT ARE BEST PRACTICES? PLEASE CRITIQUE! I think that one should at least: 1-use STRONG, random SIP passwords. Are these sent clear text across the internet? 2-Where possible one should not use auth names that match the extension number? ??? - please advise. I think one may want to: 3-Run IDS/IPS on their router. ??? - please advise. Without getting into the complexities of multi-homed, interface-specific bindings etc, are there additional precautions I should be taking? For example I tried to block registrations from other subnets as follows: [general] ... deny=0.0.0.0/0.0.0.0 ;deny all by default? permit=10.1.0.0/255.255.0.0 ;allow registrations from local subnet? But this seems to have no effect. Of course I may NOT have wanted its 'effect' if its effect would be to deny ALL SIP traffic from ALL places including my ITSP's and guest SIP URI invites. Obviously I ONLY want to disallow foreign REGISTRATIONS (from other subnets) while preserving inbound calls from ANYONE. Is there a way to do that without an SBC? For crude IPS/IDS is there an Asterisk method to blacklist registrations from a specific IP address after a certain number of failed registration attempts, or would I need an SBC or IDS/IPS for that? Thanks in advance to anyone takes a moment do a brain-dump on this topic! -Karl ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - Which rackable case for mini-ITX boards ?
From what people have said Asterisk does not require a huge amount of memory or CPU then ? I only have a couple of extensions. Running the G729 codec and will look at the Octava software for the PSTN to reduce echo. Regards, -- --[ UxBoD ]-- // PGP Key: curl -s http://www.splatnix.net/uxbod.asc | gpg --import // Fingerprint: F57A 0CBD DD19 79E9 1FCC A612 CB36 D89D 2C5A 3A84 // Keyserver: www.keyserver.net Key-ID: 0x2C5A3A84 // Phone: +44 845 869 2749 SIP Phone: [EMAIL PROTECTED] - Michael Graves [EMAIL PROTECTED] wrote: Just about anything bootable will serve you well enough. My home office is presently running of an HP T5700 thin client (1 GHz transmeta CPU, 256 MB RAM, 256 MB Flash DOM) running Astlinux. Cheap, diskless, fanless, silent, cool, reliable. Almost perfect IMHO. It's connected to a snom m3 system, 6 Polycom Aastra SIP phones, and one lowly Sipura SPA-2002. We don't have any POTS lines so no FXO card anymore. I did have a TDM400p at one point. Michael Graves -- 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - Which rackable case for mini-ITX boards ?
For a couple of years I used a Soekris Net4801 as an Astlinux host. It features a 266 MHz Geode CPU. That system was able to sucessfully transcode only two G.729 encoded calls at a time. Since moving to the 1 GHz host I've not bothered to try and overtax the transcode capability as it does as much as I need. I expect that it would handle 6-8 transcodes at a time, which is more than I need. Bear in mind that you may not need to transcode very often. If you're using primarily the PSTN through the TDM400p then why bother with G.729 at all? Michael On Sun, 24 Aug 2008 20:21:31 +0100 (BST), --[ UxBoD ]-- wrote: From what people have said Asterisk does not require a huge amount of memory or CPU then ? I only have a couple of extensions. Running the G729 codec and will look at the Octava software for the PSTN to reduce echo. Regards, -- --[ UxBoD ]-- // PGP Key: curl -s http://www.splatnix.net/uxbod.asc | gpg --import // Fingerprint: F57A 0CBD DD19 79E9 1FCC A612 CB36 D89D 2C5A 3A84 // Keyserver: www.keyserver.net Key-ID: 0x2C5A3A84 // Phone: +44 845 869 2749 SIP Phone: [EMAIL PROTECTED] - Michael Graves [EMAIL PROTECTED] wrote: Just about anything bootable will serve you well enough. My home office is presently running of an HP T5700 thin client (1 GHz transmeta CPU, 256 MB RAM, 256 MB Flash DOM) running Astlinux. Cheap, diskless, fanless, silent, cool, reliable. Almost perfect IMHO. It's connected to a snom m3 system, 6 Polycom Aastra SIP phones, and one lowly Sipura SPA-2002. We don't have any POTS lines so no FXO card anymore. I did have a TDM400p at one point. Michael Graves -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. Internal Virus Database is out of date. Checked by AVG - http://www.avg.com Version: 8.0.138 / Virus Database: 270.6.4/1617 - Release Date: 8/17/2008 12:58 PM -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - Which rackable case for mini-ITX boards ?
Reason for the G729 is that I have a couple of numbers which come in directly over IAX. Switched office numbers away from BT onto VoIP to reduce cost :) Regards, -- --[ UxBoD ]-- // PGP Key: curl -s http://www.splatnix.net/uxbod.asc | gpg --import // Fingerprint: F57A 0CBD DD19 79E9 1FCC A612 CB36 D89D 2C5A 3A84 // Keyserver: www.keyserver.net Key-ID: 0x2C5A3A84 // Phone: +44 845 869 2749 SIP Phone: [EMAIL PROTECTED] - Michael Graves [EMAIL PROTECTED] wrote: For a couple of years I used a Soekris Net4801 as an Astlinux host. It features a 266 MHz Geode CPU. That system was able to sucessfully transcode only two G.729 encoded calls at a time. Since moving to the 1 GHz host I've not bothered to try and overtax the transcode capability as it does as much as I need. I expect that it would handle 6-8 transcodes at a time, which is more than I need. Bear in mind that you may not need to transcode very often. If you're using primarily the PSTN through the TDM400p then why bother with G.729 at all? Michael -- 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SECURITY QUESTION SANITY CHECK
On Sunday 24 August 2008 14:17:47 Karl Fife wrote: For crude IPS/IDS is there an Asterisk method to blacklist registrations from a specific IP address after a certain number of failed registration attempts, or would I need an SBC or IDS/IPS for that? There is no solution in Asterisk currently, but there's an issue open that is unrelated to your question but whose solution will fix your problem: http://bugs.digium.com/view.php?id=11776 Basically, you could use this solution to forbid certain peers from registering from the Internet side of your Asterisk system, leaving only registrations from the private internal side. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] entering a password to have access to a sip account?!
Hello Steve,
thanks for the advice :)
though one prob! if i add the authenticate line itll require all callers to
enter 1234 to access *ANY* sip account..
even though this would come in handy at some point but at the moment i just
want to deny the extension 300 from being able to call 01 unless the caller
entered a password..
find below wht i did so far..
[sipura-line]
exten = 301,1,Answer() ; Answer inbound calls
exten = 301,2,Playback(silence/1)
exten = 301,3,Background(simzy1) ; input an extension
exten = 301,4,authenticate(1234)
exten = 301,5,WaitExten(8)
exten = 301,6,Dial(SIP/100,15) ; goes to operator
exten = 301,3,Wait(8)
include = spa
exten = _XXX,6,VoiceMail([EMAIL PROTECTED])
exten = 301,n,Hangup()
[spa]
exten =_301,1,GoTo(sipura-line,${EXTEN},1)
exten = _1XX,1,Dial(SIP/${EXTEN},20) ;each ring equals to 5 seconds so it will
ring 3 times
exten = _1XX,2,VoiceMail([EMAIL PROTECTED]) ; direct 2 voicemail box if line
is busy or unavailable
exten = _1XX,3,HangUp()
exten = _2XX,1,Dial(SIP/${EXTEN},20) ;each ring equals to 5 seconds so it will
ring 3 times
exten = _2XX,2,VoiceMail([EMAIL PROTECTED]) ; directs to voicemail box if line
is busy or unavailable
exten = _2XX,3,HangUp()
exten = _3XX,1,Dial(SIP/${EXTEN},20) ; each ring equals to 5 seconds so it
will ring 3 times
exten = _3XX,2,VoiceMail([EMAIL PROTECTED]) ; directs 2 voicemail box if line
is busy or unavailable
exten = _3XX,3,HangUp()
exten =_01,1,Dial(SIP/$(EXTEN)@300) ; old ogero line
;exten =_01,2,Set(TIMEOUT(absolute)=5)
exten =_02,1,Dial(SIP/$(EXTEN)@304) ; new ogero line
exten =_03,1,Dial(SIP/$(EXTEN)@305) ; samer
exten = 303,1,VoicemailMain ; voicemail box to be redirected to
Date: Sun, 24 Aug 2008 12:05:02 -0400
From: [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] entering a password to have access to a sip
account?!
You want to use Authenticate() between answer and dial.
http://www.google.com/search?q=asterisk+authenticateie=utf-8oe=utf-8aq=trls=org.mozilla:en-US:officialclient=firefox-a
Thanks,
Steve Totaro
On Sun, Aug 24, 2008 at 11:26 AM, RoLaNd RoLaNd [EMAIL PROTECTED] wrote:
Hi all,
i;m obviously a newbie, its been 2 days that im trying to figure out a way
to deny a specific extension (300) from calling another specific extensions
(03) except if the caller punch a specified password.. sorry if im not
explaining myself well.. heres an example:
i called my pstn line(with 300 as its sip account), an attendant answers and
asks me to punch in an extension number right now if i dial 03 it rings at
the other end! though i dont want that to happen! i want to set asterisk up
in a way tht if i dial 03 from 300 to ask me for a password... or it
wont let the line go through!
can anyone guide me through this issue! im really going crazy to get this
done! any help would truly and utterly be appreciated:)
ps: find below my extensions.conf
[sipura-line]
exten = 301,1,Answer() ; Answer inbound calls
exten = 301,2,Playback(silence/1)
exten = 301,3,Background(simzy1) ; input an extension
exten = 301,4,WaitExten(8)
exten = 301,5,Dial(SIP/100,15) ; goes to operator
exten = 301,4,Wait(8)
include = spa
exten = _XXX,6,VoiceMail([EMAIL PROTECTED])
exten = 301,n,Hangup()
[spa]
exten =_301,1,GoTo(sipura-line,${EXTEN},1)
exten = _1XX,1,Dial(SIP/${EXTEN},20) ;each ring equals to 5 seconds so it
will ring 3 times
exten = _1XX,2,VoiceMail([EMAIL PROTECTED]) ; direct 2 voicemail box if
line
is busy or unavailable
exten = _1XX,3,HangUp()
exten = _2XX,1,Dial(SIP/${EXTEN},20) ;each ring equals to 5 seconds so it
will ring 3 times
exten = _2XX,2,VoiceMail([EMAIL PROTECTED]) ; directs to voicemail box if
line is busy or unavailable
exten = _2XX,3,HangUp()
exten = _3XX,1,Dial(SIP/${EXTEN},20) ; each ring equals to 5 seconds so it
will ring 3 times
exten = _3XX,2,VoiceMail([EMAIL PROTECTED]) ; directs 2 voicemail box if
line is busy or unavailable
exten = _3XX,3,HangUp()
exten =_01,1,Dial(SIP/$(EXTEN)@300) ; old ogero line
;exten =_01,2,Set(TIMEOUT(absolute)=5)
exten =_02,1,Dial(SIP/$(EXTEN)@304) ; new ogero line
exten =_03,1,Dial(SIP/$(EXTEN)@305) ; samer
exten =_04,1,Dial(SIP/$(EXTEN)@306) ; gilberte
exten =_05,1,Dial(SIP/$(EXTEN)@307) ; conference
exten =_06,1,Dial(SIP/$(EXTEN)@308) ; line 4
exten = 303,1,VoicemailMain ; voicemail box to be redirected to
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Re: [asterisk-users] Global VoIP Calls?
Thanks all for your suggestions. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - Which rackable case for mini-ITX boards ?
On Sun, 24 Aug 2008, --[ UxBoD ]-- wrote: Reason for the G729 is that I have a couple of numbers which come in directly over IAX. Switched office numbers away from BT onto VoIP to reduce cost :) Don't use g729 unless you really need it. Even on an old ADSL-1 connection in the UK (256Kbps upstream), you can run 2 G711 calls over IAX (b/w needed on an IAX trunk for 2 calls is 80+65 = 145Kb/sec each way. (give or take) Even 2 SIP calls @ 160Kbps ought to be fine. However I've transcoded over 10 concurrent g729 calls on a 1GHz VIA processor, so it's possible if you really need to do it. Gordon Regards, -- --[ UxBoD ]-- // PGP Key: curl -s http://www.splatnix.net/uxbod.asc | gpg --import // Fingerprint: F57A 0CBD DD19 79E9 1FCC A612 CB36 D89D 2C5A 3A84 // Keyserver: www.keyserver.net Key-ID: 0x2C5A3A84 // Phone: +44 845 869 2749 SIP Phone: [EMAIL PROTECTED] - Michael Graves [EMAIL PROTECTED] wrote: For a couple of years I used a Soekris Net4801 as an Astlinux host. It features a 266 MHz Geode CPU. That system was able to sucessfully transcode only two G.729 encoded calls at a time. Since moving to the 1 GHz host I've not bothered to try and overtax the transcode capability as it does as much as I need. I expect that it would handle 6-8 transcodes at a time, which is more than I need. Bear in mind that you may not need to transcode very often. If you're using primarily the PSTN through the TDM400p then why bother with G.729 at all? Michael -- 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problems with D-channel (PRI)
It was driver error caused disk failure. It is fixed now but we still have problems with D channel up/down messages. Can we fix it reconfiguring software or changing something? - Original Message - From: Rob Hillis [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, August 23, 2008 12:31 PM Subject: Re: [asterisk-users] Problems with D-channel (PRI) Jakub Arkon Syrek wrote: Hello, we have strange problem, till now everything was working fine, there where no problems with dial and answer calls. Yesterday our system crashed and we notice strange behavior. What type of event caused the box to crash? Given the fact that you've also mentioned a degraded RAID array, this is sounding very much like a power spike that may have damaged hardware. What type of card is installed in your machine? Have you spoken to the relevant support people for that card? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] entering a password to have access to a sip account?!
I have one solution in mind, maybe it is an overkill but:
You can create a db entry for each sip account, DB(family/key) lets name
family=destination sip number and key=${Callerid(num)} and assing a value 0
or 1, so string will be like this DB(301/300)=1 fot that 300 sip account,
and for all other sip accounts DB(300/NNN)=0 where NNN are all others sip
accounts numbers. You can use set for this, example
exten = 75,1,Set(DB(300/301)=1)
or
exten = 75,1,Set(DB(300/${Callerid(num)}=1)
exten = 76,1,Set(DB(300/${Callerid(num)}=0)
And just go and call from each phone 75 or 76 , i assume that you callerid
is the same as callerid(num) var. The methos is somehow primitive and will
not work if you have 500 extensions, but for 5 sip accounts is a way to go.
Or create external bash script to speed up.
After this you will have as much db entryes as sip accounts in you astdb,
all we need to is is to verify the value before call
exten = 300,1,GotoIf($[${DB(300/${Callerid(num)})}=1]?2:3)
exten = 300,2,Playback(stop_calling_me)
exten = 300,3,Dial(Sip/300)
And again i assume that your sip peers have the same
Callerid(num)=extensions
Maybe i got some syntax errors, but you get the idea.
Have fun
2008/8/24 RoLaNd RoLaNd [EMAIL PROTECTED]
Hello Steve,
thanks for the advice :)
though one prob! if i add the authenticate line itll require all callers to
enter 1234 to access *ANY* sip account..
even though this would come in handy at some point but at the moment i
just want to deny the extension 300 from being able to call 01 unless the
caller entered a password..
find below wht i did so far..
[sipura-line]
exten = 301,1,Answer() ; Answer inbound calls
exten = 301,2,Playback(silence/1)
exten = 301,3,Background(simzy1) ; input an extension
exten = 301,4,authenticate(1234)
exten = 301,5,WaitExten(8)
exten = 301,6,Dial(SIP/100,15) ; goes to operator
exten = 301,3,Wait(8)
include = spa
exten = _XXX,6,VoiceMail([EMAIL PROTECTED])
exten = 301,n,Hangup()
[spa]
exten =_301,1,GoTo(sipura-line,${EXTEN},1)
exten = _1XX,1,Dial(SIP/${EXTEN},20) ;each ring equals to 5 seconds so it
will ring 3 times
exten = _1XX,2,VoiceMail([EMAIL PROTECTED]) ; direct 2 voicemail box if
line is busy or unavailable
exten = _1XX,3,HangUp()
exten = _2XX,1,Dial(SIP/${EXTEN},20) ;each ring equals to 5 seconds so it
will ring 3 times
exten = _2XX,2,VoiceMail([EMAIL PROTECTED]) ; directs to voicemail box if
line is busy or unavailable
exten = _2XX,3,HangUp()
exten = _3XX,1,Dial(SIP/${EXTEN},20) ; each ring equals to 5 seconds so it
will ring 3 times
exten = _3XX,2,VoiceMail([EMAIL PROTECTED]) ; directs 2 voicemail box if
line is busy or unavailable
exten = _3XX,3,HangUp()
exten =_01,1,Dial(SIP/$(EXTEN)@300) ; old ogero line
;exten =_01,2,Set(TIMEOUT(absolute)=5)
exten =_02,1,Dial(SIP/$(EXTEN)@304) ; new ogero line
exten =_03,1,Dial(SIP/$(EXTEN)@305) ; samer
exten = 303,1,VoicemailMain ; voicemail box to be redirected to
Date: Sun, 24 Aug 2008 12:05:02 -0400
From: [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] entering a password to have access to a sip
account?!
You want to use Authenticate() between answer and dial.
http://www.google.com/search?q=asterisk+authenticateie=utf-8oe=utf-8aq=trls=org.mozilla:en-US:officialclient=firefox-a
Thanks,
Steve Totaro
On Sun, Aug 24, 2008 at 11:26 AM, RoLaNd RoLaNd [EMAIL PROTECTED]
wrote:
Hi all,
i;m obviously a newbie, its been 2 days that im trying to figure out a
way
to deny a specific extension (300) from calling another specific
extensions
(03) except if the caller punch a specified password.. sorry if im not
explaining myself well.. heres an example:
i called my pstn line(with 300 as its sip account), an attendant
answers and
asks me to punch in an extension number right now if i dial 03 it
rings at
the other end! though i dont want that to happen! i want to set
asterisk up
in a way tht if i dial 03 from 300 to ask me for a password... or
it
wont let the line go through!
can anyone guide me through this issue! im really going crazy to get
this
done! any help would truly and utterly be appreciated:)
ps: find below my extensions.conf
[sipura-line]
exten = 301,1,Answer() ; Answer inbound calls
exten = 301,2,Playback(silence/1)
exten = 301,3,Background(simzy1) ; input an extension
exten = 301,4,WaitExten(8)
exten = 301,5,Dial(SIP/100,15) ; goes to operator
exten = 301,4,Wait(8)
include = spa
exten = _XXX,6,VoiceMail([EMAIL PROTECTED])
exten = 301,n,Hangup()
[spa]
exten =_301,1,GoTo(sipura-line,${EXTEN},1)
exten = _1XX,1,Dial(SIP/${EXTEN},20) ;each ring equals to 5 seconds so
it
will ring 3 times
exten = _1XX,2,VoiceMail([EMAIL PROTECTED]) ; direct 2 voicemail box
if line
is busy or unavailable
exten = _1XX,3,HangUp()
exten =
Re: [asterisk-users] entering a password to have access to a sip account?!
I have one solution in mind, maybe it is an overkill but:
You can create a db entry for each sip account, DB(family/key) lets name
family=destination sip number and key=${Callerid(num)} and assing a value 0
or 1, so string will be like this DB(301/300)=1 fot that 300 sip account,
and for all other sip accounts DB(300/NNN)=0 where NNN are all others sip
accounts numbers. You can use set for this, example
exten = 75,1,Set(DB(300/301)=1)
or
exten = 75,1,Set(DB(300/${Callerid(num)}=1)
exten = 76,1,Set(DB(300/${Callerid(num)}=0)
And just go and call from each phone 75 or 76 , i assume that you callerid
is the same as callerid(num) var. The methos is somehow primitive and will
not work if you have 500 extensions, but for 5 sip accounts is a way to go.
Or create external bash script to speed up.
After this you will have as much db entryes as sip accounts in you astdb,
all we need to is is to verify the value before call
exten = 300,1,GotoIf($[${DB(300/${Callerid(num)})}=1]?2:3)
exten = 300,2,Playback(stop_calling_me)
exten = 300,3,Dial(Sip/300)
And again i assume that your sip peers have the same
Callerid(num)=extensions
Maybe i got some syntax errors, but you get the idea.
Have fun
previous message have failed for some reasons.
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[asterisk-users] RemoveQueueMember race condition
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi all, I've got a problem with RemoveQueueMember. If an agent receives a call at the same time that they're being removed from the queue, they receive that call and remain in to the queue (and will receive calls until they're able to be removed). The logic works when the call doesn't barge in on them. I guess that pausing the agent before the remove would just transfer the problem to the pausing. Any suggestions? Regards, Paul - -- Paul Crane Technical Support Officer VentureVoIP Ltd John Wickliffe House 265 Princes Street Dunedin Phone: +64 3 951 3107 Web: www.venturevoip.com MSN: [EMAIL PROTECTED] ICQ: 381715372 -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (Darwin) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFIse3decPQQzzU6hQRAqIIAJ9BbrbX15jwlMcnaujK+9SW92UPswCfcH2z GcDr1IL+viLn9bCrmbPNi/I= =q+5l -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] entering a password to have access to a sip account?!
Roland,
The simple solution is to utilize the power of contexts (put exten 300
in a different context in sip.conf or db) and includes to separate yet
include 300 (so 300 can be called and call other internal extensions).
Add authenticate before the dial statement.
The easiest way to do it, is just copy the [spa] context in your
dialplan and then change the context from [spa] to [restricted_300]
(or whatever) and then just add the authenticate statement as below
and renumber the dial prio to 2 or (n for next). Make sure your
context in sip.conf for that sip extension matches this newly created
context. There are probably cleaner ways of doing it, but one thing
at a time :)
exten =_01,1,Authenticate(1234)
exten =_01,2,Dial(SIP/$(EXTEN)@300) ; old ogero line
Thanks,
Steve Totaro
On Sun, Aug 24, 2008 at 4:20 PM, RoLaNd RoLaNd [EMAIL PROTECTED] wrote:
Hello Steve,
thanks for the advice :)
though one prob! if i add the authenticate line itll require all callers to
enter 1234 to access *ANY* sip account..
even though this would come in handy at some point but at the moment i just
want to deny the extension 300 from being able to call 01 unless the
caller entered a password..
find below wht i did so far..
[sipura-line]
exten = 301,1,Answer() ; Answer inbound calls
exten = 301,2,Playback(silence/1)
exten = 301,3,Background(simzy1) ; input an extension
exten = 301,4,authenticate(1234)
exten = 301,5,WaitExten(8)
exten = 301,6,Dial(SIP/100,15) ; goes to operator
exten = 301,3,Wait(8)
include = spa
exten = _XXX,6,VoiceMail([EMAIL PROTECTED])
exten = 301,n,Hangup()
[spa]
exten =_301,1,GoTo(sipura-line,${EXTEN},1)
exten = _1XX,1,Dial(SIP/${EXTEN},20) ;each ring equals to 5 seconds so it
will ring 3 times
exten = _1XX,2,VoiceMail([EMAIL PROTECTED]) ; direct 2 voicemail box if line
is busy or unavailable
exten = _1XX,3,HangUp()
exten = _2XX,1,Dial(SIP/${EXTEN},20) ;each ring equals to 5 seconds so it
will ring 3 times
exten = _2XX,2,VoiceMail([EMAIL PROTECTED]) ; directs to voicemail box if
line is busy or unavailable
exten = _2XX,3,HangUp()
exten = _3XX,1,Dial(SIP/${EXTEN},20) ; each ring equals to 5 seconds so it
will ring 3 times
exten = _3XX,2,VoiceMail([EMAIL PROTECTED]) ; directs 2 voicemail box if
line is busy or unavailable
exten = _3XX,3,HangUp()
exten =_01,1,Dial(SIP/$(EXTEN)@300) ; old ogero line
;exten =_01,2,Set(TIMEOUT(absolute)=5)
exten =_02,1,Dial(SIP/$(EXTEN)@304) ; new ogero line
exten =_03,1,Dial(SIP/$(EXTEN)@305) ; samer
exten = 303,1,VoicemailMain ; voicemail box to be redirected to
Date: Sun, 24 Aug 2008 12:05:02 -0400
From: [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] entering a password to have access to a sip
account?!
You want to use Authenticate() between answer and dial.
http://www.google.com/search?q=asterisk+authenticateie=utf-8oe=utf-8aq=trls=org.mozilla:en-US:officialclient=firefox-a
Thanks,
Steve Totaro
On Sun, Aug 24, 2008 at 11:26 AM, RoLaNd RoLaNd [EMAIL PROTECTED]
wrote:
Hi all,
i;m obviously a newbie, its been 2 days that im trying to figure out a
way
to deny a specific extension (300) from calling another specific
extensions
(03) except if the caller punch a specified password.. sorry if im not
explaining myself well.. heres an example:
i called my pstn line(with 300 as its sip account), an attendant answers
and
asks me to punch in an extension number right now if i dial 03 it
rings at
the other end! though i dont want that to happen! i want to set asterisk
up
in a way tht if i dial 03 from 300 to ask me for a password... or it
wont let the line go through!
can anyone guide me through this issue! im really going crazy to get
this
done! any help would truly and utterly be appreciated:)
ps: find below my extensions.conf
[sipura-line]
exten = 301,1,Answer() ; Answer inbound calls
exten = 301,2,Playback(silence/1)
exten = 301,3,Background(simzy1) ; input an extension
exten = 301,4,WaitExten(8)
exten = 301,5,Dial(SIP/100,15) ; goes to operator
exten = 301,4,Wait(8)
include = spa
exten = _XXX,6,VoiceMail([EMAIL PROTECTED])
exten = 301,n,Hangup()
[spa]
exten =_301,1,GoTo(sipura-line,${EXTEN},1)
exten = _1XX,1,Dial(SIP/${EXTEN},20) ;each ring equals to 5 seconds so
it
will ring 3 times
exten = _1XX,2,VoiceMail([EMAIL PROTECTED]) ; direct 2 voicemail box if
line
is busy or unavailable
exten = _1XX,3,HangUp()
exten = _2XX,1,Dial(SIP/${EXTEN},20) ;each ring equals to 5 seconds so
it
will ring 3 times
exten = _2XX,2,VoiceMail([EMAIL PROTECTED]) ; directs to voicemail box
if
line is busy or unavailable
exten = _2XX,3,HangUp()
exten = _3XX,1,Dial(SIP/${EXTEN},20) ; each ring equals to 5 seconds so
it
will ring 3 times
exten = _3XX,2,VoiceMail([EMAIL PROTECTED]) ; directs 2 voicemail box if
line is busy or unavailable
exten = _3XX,3,HangUp()
exten
[asterisk-users] wct4xxp alarmdebounce
Anyone has tried wct4xxp drivers' alarmdebounce parameter ? I search the internet no one seems to have used it, is this how the parameter can be specified, eg :- # modprobe wct4xxp alarmdebounce=200 Does it have the effect of making asterisk PRI more tolerant with poorer quality lines ? Regards. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MWI working perfectly. Shouldn't it be broken??
I see. Thanks Russel. And I now notice that if I explicitly tell my 962 the IP address of the mail server, it will also subscribe. Do I understand correctly that we are not talking about redundant MWI status traffic here, we're ONLY talking about the notion that asterisk ignores MWI subscription status and behaves as if it has 100% MWI subscription. That is unless subscribemwi=yes is in sip.conf. Is that an accurate summary? And theoretically, if I had thousands of endpoints and I needed 100% mwi subscription, there may be some theoretical efficiency to turning off all MWI subscriptions in all of the endpoints. Likewise if only 25% of my 'thousands' of endpoints needed any MWI, there would be some efficiency in setting subscribemwi=yes, and explicitly subscribing only those 25%. Is that right? Thanks again for clarifying! I appreciate it! -Karl On Sun, 24 Aug 2008 11:50:19 -0500, Russell Bryant [EMAIL PROTECTED] said: On Aug 24, 2008, at 11:29 AM, Karl Fife wrote: FYI, it's not an issue of the subscription not YET subscribing etc. If I were to restart the system and endpoints, all the subs slowly show up one by one, but the 962 never does -- even after days, weeks, and months. Yet the MWI always works perfectly from the get-go. Asterisk will send the NOTIFY for MWI even if the device doesn't subscribe, unless you tell it not to. This is necessary for some phones for MWI to work. If you _don't_ want Asterisk to do this, you can set the subscribemwi=yes option in sip.conf. This tells Asterisk to _only_ send MWI with an associated subscription. -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
