Re: [asterisk-users] Astribank problem
I do some test: 1.unplug usb connector from server to astricon 2.unplug power to astricon 3.plug-in the power to astricon 4.plug-in the usb connector Here is the log from /var/log/messages after doing the 1st step. Feb 1 19:38:24 localhost last message repeated 2 times Feb 1 19:43:39 localhost kernel: ERR-xpp_usb: xusb-0 (usb-:00:1d.7-3) [X1038295]: nonzero write bulk status received: -71 (pending_writes=1) Feb 1 19:43:39 localhost kernel: usb 2-3: USB disconnect, address 3 Feb 1 19:43:39 localhost kernel: ERR-xpp_usb: XBUS-00: xusb_listen: usb_submit_urb failed: -19 Feb 1 19:43:39 localhost kernel: INFO-xpp: XBUS-00: [usb:X1038295] Disconnecting Feb 1 19:43:39 localhost kernel: INFO-xpp: XBUS-00: [usb:X1038295] Deactivating Feb 1 19:43:39 localhost kernel: INFO-xpp: XBUS-00: [usb:X1038295] Release XPDS Feb 1 19:43:39 localhost kernel: INFO-xpp: XBUS-00/XPD-00: Remove Feb 1 19:43:39 localhost kernel: INFO-xpp: XBUS-00/XPD-10: Remove Feb 1 19:43:39 localhost kernel: INFO-xpp: XBUS-00/XPD-20: Remove Feb 1 19:43:39 localhost kernel: NOTICE-xpp: worker_reset: worker(XBUS-00)->xpds_init_done=0 Feb 1 19:43:39 localhost kernel: INFO-xpp: XBUS-00: [usb:X1038295] Atribank Remove Feb 1 19:43:39 localhost kernel: INFO-xpp: XBUS-00: [usb:X1038295] Astribank Release Feb 1 19:43:39 localhost kernel: INFO-xpp_usb: xusb-0 (usb-:00:1d.7-3) [X1038295]: now disconnected Feb 1 19:43:39 localhost 'astribank_hook'[3728]: offline(XBUS-00): 0/1 from /etc/dahdi/xpp_order Feb 1 19:43:39 localhost 'astribank_hook'[3735]: All Astribanks offline And, this is the log after doing 4th step. Feb 1 19:44:20 localhost kernel: usb 2-3: new high speed USB device using ehci_hcd and address 4 Feb 1 19:44:20 localhost kernel: usb 2-3: configuration #1 chosen from 1 choice Feb 1 19:44:21 localhost 'xpp_fxloader'[3847]: Exiting... XPP_HOTPLUG_DISABLED lsusb result is: [r...@localhost ~]# lsusb Bus 002 Device 004: ID e4e4:1160 Bus 002 Device 001: ID : Bus 006 Device 001: ID : Bus 006 Device 002: ID 04b3:3025 IBM Corp. Bus 004 Device 001: ID : Bus 008 Device 001: ID : Bus 007 Device 001: ID : Bus 001 Device 001: ID : Bus 005 Device 001: ID : Bus 003 Device 001: ID : here is the msg when i do /usr/share/dahdi/xpp_fxloader [r...@localhost ~]# /usr/share/dahdi/xpp_fxloader usb 'xpp_fxloader'[3955]: - FIRMWARE LOADING: (usb) [1 devices] Got all 1 devices 'xpp_fxloader'[4074]: - FIRMWARE IS LOADED but, when i do /etc/init.d/dahdi stop and start here is the result [r...@localhost ~]# /etc/init.d/dahdi start Loading DAHDI hardware modules: xpp_usb: [ OK ] Astribanks initialization is starting Astribanks detection ..TIMEOUT No hardware timing source found in /proc/dahdi, loading dahdi_dummy Running dahdi_cfg: DAHDI_CHANCONFIG failed on channel 1: No such device or address (6) [FAILED] On Sat, Jan 30, 2010 at 03:57:30AM +, frangky robert wrote: > > H all... > > I have an Astribank (8FXS/16FXO), IBM X3200 M2, Asterisk-1.6.2.1, > dahdi-linux-complete-2.2.1, libpri-1.4.10.2, centos-5.4 final. > My problem is, every time i unplug the astribank power supply, and > reconnect it, astribank cannot work again (lsusb result is 11x0)... When you plug the Astribank, the firmware should get loaded by a script called from udev. What messages do you see in /var/log/messages following that? > but, after reinstall the asterisk and dahdi, astribank will detected > (lsusb result is 11x2)... -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir _ Windows Live: Friends get your Flickr, Yelp, and Digg updates when they e-mail you. http://www.microsoft.com/windows/windowslive/see-it-in-action/social-network-basics.aspx?ocid=PID23461::T:WLMTAGL:ON:WL:en-id:SI_SB_3:092010-- _ -- 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] connect problem unless when verbose
hi all, just had a terrible and sleepless weekend at the office trying to get asterisk going, its just tough love ;) have tried several asterisk versions but i currently have the following setup on debian lenny that kind of works. asterisk-1.6.2.0 dahdi-linux-complete-2.2.0.2+2.2.0 libpri-1.4.10.2 freepbx-2.6.0 setting up the sip devices is no problem at all, the difficulty i have is setting up 6xisdn2 lines with 2xb410p cards. besides the fact that i have no clue about what i'm doing i find the available documentation very very confusing, but i finally managed to make outgoing calls to my mobile this morning, sort off. when calling my mobile i hear a ringtone on my sip device and my mobile actually rings, YEAH!!! however, when i accept the call on my mobile my sip device keeps on ringing and my mobile gives no sound at all, when cancelling the call it simply cancels. except, and this i don't understand, i issue asterisk -rv (only with the v option), then i can connect and talk to myself, i often talk to myself when i spent a weekend at the office but this time its justifiable ;) anybody has a clue what could trigger this behavior??? -- _ -- 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] FCT for 3G Video calls
Hi Is anyone aware of a fixed cellular terminal that supports 3G video calls? Regards, Chris -- _ -- 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] Odd error mssage on DAHDI lines
What's this: -- Attempting call on DAHDI/g1/9 for application Wait(5) (Retry 1) -- Requested transfer capability: 0x00 - SPEECH -- Channel 0/2, span 1 got hangup, cause 44 -- Forcing restart of channel 0/2 on span 1 since channel reported in use -- Hungup 'DAHDI/2-1' Where can I look up "cause 44". And if this is the sort of transient error that seems to be implied by the "Forcing restart" message, why isn't it retried? -- _ -- 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] app_directory broken in 1.6
Thanks Tilghman, that made a substantial difference. On Sun, Jan 31, 2010 at 6:18 PM, Tilghman Lesher wrote: > On Sunday 31 January 2010 18:12:15 cjwstudios wrote: > > Hello, > > > > I have separate contexts defined in voicemail.conf as follows: > > > > [abcdental] > > 100 => 1234,John Doe > > > > And call application directory using the following syntax: > > exten => 1,1,Directory(abcdental[,abcdental,f]) > > Uh, the square brackets in the help message means that that part > is optional. Including the square bracket literally is probably why you're > getting bad results. Try just: > > Directory(abcdental,abcdental,f) > > -- > Tilghman Lesher > Digium, Inc. | Senior Software Developer > twitter: Corydon76 | IRC: Corydon76-dig (Freenode) > 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 > -- _ -- 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] app_directory broken in 1.6
On Sunday 31 January 2010 18:12:15 cjwstudios wrote: > Hello, > > I have separate contexts defined in voicemail.conf as follows: > > [abcdental] > 100 => 1234,John Doe > > And call application directory using the following syntax: > exten => 1,1,Directory(abcdental[,abcdental,f]) Uh, the square brackets in the help message means that that part is optional. Including the square bracket literally is probably why you're getting bad results. Try just: Directory(abcdental,abcdental,f) -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) 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] sip to dahdi and billsec
Uros Djokic wrote:
> Hi,
>
> My costumers are logged in on my Asterisk PBX through XLite Softphone
> (SIP). My server is
> connected to PSTN. Problem is when SIP phone calls ordinary phone via
> dahdi I get
> DAHDI/1-1 ANSWERED SIP/number-number and billsec field from cdr is
> start counting.
>
> Is it normal behavior ? Can I change that ?
>
> So channel gets in ANSWERED state and billsec starts as soon as line
> starts
> to ring even if no one really pick up ordinary phone and costumer did
> not talk to anyone.
> That leads to problem that costumers will be billed even if they did
> not make a real
> conversation.
>
> How can I avoid that behavior and set asterisk to start counting
> billsecs after
> someone really pick up the phone on the other side ?
>
> How can I distinguish real (talking to) call from just ring (no real
> answer call)
> when both are in state ANSWERED ?
>
> I tried with timeout 20 in Dial command but since channel is
> "answered" when it
> starts to ring timeout is not doing what I want.
>
> Here is my Dial command:
> exten => _X.,n,Dial(dahdi/g0/${EXTEN},20,L(${Limit}:6:2)hH)
>
> It works very good in case ordinary phone calls sip (for incoming
> calls from PSTN)
> because I need to click answer on xlite to move call in state ANSWERED
> so if I don't
> click it is not answered and timeout works.
>
> Can you help me with that ?
>
> Thanks,
> Uros
>
>
> --
> Use Free Software http://www.fsf.org/
> ---
> Four essential software freedoms:
> 1) To study source code
> 2) To copy program
> 3) To modify source code
> 4) To redistribute modified program under condition that new user has
> all 4 freedoms.
> Richard M. Stallman
It entirely depends on the technology used to interface to the PSTN.
You have not specified what technology/hardware you are using to connect
to the PSTN.
For instance if you are using POTS(plain old telephone service - analog
copper fed lines), you do not get answer supervision back from the telco.
Lyle Giese
LCR Computer Services, Inc.
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Re: [asterisk-users] SIP Registration Failure Logging
Try: core set verbose 4 >From the Asterisk CLI -uzzi PS: If you're not seeing any connection information, be sure to double-check the IP address is correct. Learned that lesson the hard way =\ On Sun, Jan 31, 2010 at 5:51 PM, Jim Rosenberg wrote: > Let's say I have two Asterisk boxes, A and B. I am trying to get A to do > SIP registration on B, so an extension for A can dial SIP phones covered by > B. If I examine the logs on B, if the registration succeeds, I am seeing a > notice to that effect on B. But if the registration *fails*, i'm not seeing > any message logged on B. Maybe I'm just not looking in the right place. Is > there a way to turn on logging or debugging so registration failures are > logged on the "target"? > > I'm doing something profoundly stupid, and seeing the notorious > > chan_sip.c:12009 handle_response_invite: Failed to authenticate on INVITE > > message, and trying to trace why. > > -Thanks, Jim > > -- > _ > -- 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] app_directory broken in 1.6
Hello, I have separate contexts defined in voicemail.conf as follows: [abcdental] 100 => 1234,John Doe And call application directory using the following syntax: exten => 1,1,Directory(abcdental[,abcdental,f]) However since I migrated from 1.4 to 1.6, app_directory no longer parses the voicemail.conf to match the full name of the mailbox. App_directory only matches directory names based on the entries registered to the default context (voicemail show users) Therefore my workaround has been to define the users in users.conf; however the problem with that is that the fullnames in users.conf only register to the default, which means that users from multiple contexts are returned when only the specified context should be returned. It seems that the first argument of the app directory "vm-context" Directory([vm-context][,dial-context[,options]]) is broken as app_directory will only return matches from the default context regardless of vm-context specified. Any thoughts are appreciated. -- _ -- 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] callerid not working over sip
On 31 Jan 2010, at 23:17, sean darcy wrote: > Doh. It appears I was making it up. > > Thanks. No problem. If that doesn't work, try a sip debug and see what's in that. 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] callerid not working over sip
Steve Howes wrote:
> On 31 Jan 2010, at 16:24, sean darcy wrote:
>>> -- Executing [...@internal:3] Set("DAHDI/1-1", "CALLERID="Test"
>>> <447>") in new stack
>>>
>>> Why isn't the office asterisk picking up the callerid from the home
>>> asterisk?
>
> You're making up the syntax?
>
> http://www.voip-info.org/wiki/view/Setting+Callerid
>
> S
>
Doh. It appears I was making it up.
Thanks.
sean
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Re: [asterisk-users] MATH
On 31/01/10 6:27 PM, Thomas Perron wrote:
> what is wrong with this please:
>
> ;exten => 4,1,WaitExten(3)
> exten => 4,1,Set(TOTAL=${MATH(${TOTAL}+500,int)})
> exten => 4,n,WaitExten(3)
> exten => 2,1,Set(TOTAL=${MATH(${TOTAL}+200,int)})
> exten => 2,n,Waitexten(3)
> exten => 3,1,Set(TOTAL=${MATH(${TOTAL}+300,int)})
> exten => 3,n,WaitExten(3)
> exten => 9,1,SayNumber(${TOTAL})
Heh, you might need to say what you're expecting and what you're getting :D
Straight off, all I can see is that 2 does 200, 3 does 300 and 4 does 500.
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Re: [asterisk-users] callerid not working over sip
On 31 Jan 2010, at 16:24, sean darcy wrote:
>> -- Executing [...@internal:3] Set("DAHDI/1-1", "CALLERID="Test"
>> <447>") in new stack
>>
>> Why isn't the office asterisk picking up the callerid from the home
>> asterisk?
You're making up the syntax?
http://www.voip-info.org/wiki/view/Setting+Callerid
S
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[asterisk-users] SIP Registration Failure Logging
Let's say I have two Asterisk boxes, A and B. I am trying to get A to do SIP registration on B, so an extension for A can dial SIP phones covered by B. If I examine the logs on B, if the registration succeeds, I am seeing a notice to that effect on B. But if the registration *fails*, i'm not seeing any message logged on B. Maybe I'm just not looking in the right place. Is there a way to turn on logging or debugging so registration failures are logged on the "target"? I'm doing something profoundly stupid, and seeing the notorious chan_sip.c:12009 handle_response_invite: Failed to authenticate on INVITE message, and trying to trace why. -Thanks, Jim -- _ -- 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] request for testing: MixMonitor Mute
I have uploaded a patch for 1.4 and trunk that allows you to mute either or both parts of a mixmonitor recording. I would appreciate it if someone apart from me could test it and let me know how you get on. Thanks! Julian https://issues.asterisk.org/view.php?id=16740 for PCI-DSS compliance we are not allowed to record a credit card number is a MixMonitor file. However, we must record all conversations I have added a new feature to audiohooks so that you can mute either read / write or both types of frames - this allows for MixMonitor to mute either side of the conversation without affecting the conversation itself. MixMonitor now has two manager commands 1) manager show command MuteMixMonitor Action: MuteMixMonitor Synopsis: Mute a channel in MixMonitor Privilege: Description: Mutes a Mixmonitor Channel. Variables: Channel: Channel to mute. Direction: Which part to mute. read|write|both (from channel|to channel|both channels). 2) manager show command unMuteMixMonitor Action: unMuteMixMonitor Synopsis: unMute a channel in MixMonitor Privilege: Description: unMutes a Mixmonitor Channel. Variables: Channel: Channel to unmute. Direction: Which part to unmute. read|write|both (from channel|to channel|both channels). -- _ -- 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] MATH
Clue, If a caller keys in "4 5 3" will some variable return "453"? I ASSume yes, since you can make menu selections with DTMF, "obviously" you can process the results further or in other ways than that. Cary -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards Sent: Sunday, January 31, 2010 2:29 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] MATH On Sun, 31 Jan 2010, Thomas Perron wrote: > does dtmf any any variable that i can capture and use w/ some logic > like in the case of a gotoif Anyone have a clue what this means? Anyone? Anyone? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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] MATH
> > does dtmf any any variable that i can capture and use w/ some logic like > > in the case of a gotoif > > Anyone have a clue what this means? Anyone? Anyone? How about this: "does dtmf transmit any variable that i can capture and use w/ some logic like [in the case of a] gotoif" Philipp -- _ -- 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] sip to dahdi and billsec
Hi,
My costumers are logged in on my Asterisk PBX through XLite Softphone (SIP).
My server is
connected to PSTN. Problem is when SIP phone calls ordinary phone via dahdi
I get
DAHDI/1-1 ANSWERED SIP/number-number and billsec field from cdr is start
counting.
Is it normal behavior ? Can I change that ?
So channel gets in ANSWERED state and billsec starts as soon as line starts
to ring even if no one really pick up ordinary phone and costumer did not
talk to anyone.
That leads to problem that costumers will be billed even if they did not
make a real
conversation.
How can I avoid that behavior and set asterisk to start counting billsecs
after
someone really pick up the phone on the other side ?
How can I distinguish real (talking to) call from just ring (no real answer
call)
when both are in state ANSWERED ?
I tried with timeout 20 in Dial command but since channel is "answered" when
it
starts to ring timeout is not doing what I want.
Here is my Dial command:
exten => _X.,n,Dial(dahdi/g0/${EXTEN},20,L(${Limit}:6:2)hH)
It works very good in case ordinary phone calls sip (for incoming calls from
PSTN)
because I need to click answer on xlite to move call in state ANSWERED so if
I don't
click it is not answered and timeout works.
Can you help me with that ?
Thanks,
Uros
--
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---
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1) To study source code
2) To copy program
3) To modify source code
4) To redistribute modified program under condition that new user has all 4
freedoms.
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Re: [asterisk-users] asterisk-users Digest, Vol 66, Issue 75
Hi Shahnawaz Have you considered how you are going to address location issue for Mobile users calling 911. You should think of SS7 MAP/TCAP to atleast know their Cell ID Regards Sam > Thanks very much everybody who contributed their thoughts. I would try > to get some DID's so that each physical location can be identified by > 911 call Center. > > Regards > > Shahnawaz > > On 2010-01-29, at 2:41 PM, Kevin P. Fleming wrote: > >> Leif Neland wrote: >> >>> 2: Often callers are answered with an automated message "This is 911, >>> please hold", just to give pranksters/misdiallers a chance to hang up >>> before "disturbing" the operator. Unless 911 records the incoming >>> call >>> right from the start, they will never hear the "im-at" message. And >>> even >>> if they do, they have to know the message is there to seek on the >>> recording. >> >> In the US at least, calls to PSAPs are recorded from the instant the >> last digit is dialed, before the call is even routed and ringing (on >> wireline networks where this is possible, anyway). >> >> -- >> 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] 911, Location
Hi Shahnawaz Have you considered how you are going to address location issue for Mobile users calling 911. You should think of SS7 MAP/TCAP to atleast know their Cell ID Regards Sam > Thanks very much everybody who contributed their thoughts. I would try to get some DID's so that each physical location can be identified by 911 call Center. > > Regards > > Shahnawaz > > On 2010-01-29, at 2:41 PM, Kevin P. Fleming wrote: > >> Leif Neland wrote: >> >>> 2: Often callers are answered with an automated message "This is 911, please hold", just to give pranksters/misdiallers a chance to hang up before "disturbing" the operator. Unless 911 records the incoming call >>> right from the start, they will never hear the "im-at" message. And even >>> if they do, they have to know the message is there to seek on the recording. >> >> In the US at least, calls to PSAPs are recorded from the instant the last digit is dialed, before the call is even routed and ringing (on wireline networks where this is possible, anyway). >> >> -- >> 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] MATH
On Sun, 31 Jan 2010, Thomas Perron wrote: > does dtmf any any variable that i can capture and use w/ some logic > like in the case of a gotoif Anyone have a clue what this means? Anyone? Anyone? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MATH
does dtmf any any variable that i can capture and use w/ some logic
like in the case of a gotoif
so, if caller enters a certain number then gotoif matches XX
otherwise go to YY.
On Sun, Jan 31, 2010 at 10:58 AM, Thomas Perron wrote:
> ok.
> that worked
> thanks!!
>
>
> On Sun, Jan 31, 2010 at 10:50 AM, Tzafrir Cohen
> wrote:
>> On Sun, Jan 31, 2010 at 10:37:29AM -0500, Thomas Perron wrote:
>>> hi
>>> i don't claim to be a star at this but there must be some obvious part
>>> missing;
>>> my dial plan is below. out put from cli follows.
>>>
>>> exten => 3011,1,Answer()
>>> exten => 3011,n,Set(TOTAL=0)
>>> exten => 3011,n,Set(TOTAL=${Math(${TOTAL}+300,int)})
>>> exten => 3011,n,WaitExten(3)
>>> exten => 988,1,SayNumber(${TOTAL})
>>>
>>> [Jan 31 10:21:35] ERROR[1318]: pbx.c:2770 ast_func_read: Function Math
>>> not registered
>>
>> Function names are CaSe SenSitive, and are normally ALL CAPS. You should
>> use 'MATH' instead of 'Math'.
>>
>> /me is done shouting for today, hopefully.
>>
>> --
>> Tzafrir Cohen
>> icq#16849755 jabber:tzafrir.co...@xorcom.com
>> +972-50-7952406 mailto:tzafrir.co...@xorcom.com
>> http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
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>> To UNSUBSCRIBE or update options visit:
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
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Re: [asterisk-users] callerid not working over sip
sean darcy wrote:
> Calling from my home using Asterisk 1.6.2.1 to an office extension
> (Asterisk 1.6.1.13) the callerid is not honored:
>
> Home:
>
> -- Starting simple switch on 'DAHDI/1-1'
> -- Executing [...@internal:1] Answer("DAHDI/1-1", "") in new stack
> -- Executing [...@internal:2] NoOp("DAHDI/1-1", "Context:
> office-extensions") in new stack
> -- Executing [...@internal:3] Set("DAHDI/1-1", "CALLERID="Test"
> <447>") in new stack
> -- Executing [...@internal:4] Dial("DAHDI/1-1",
> "SIP/office-home-sip/170") in new stack
>== Using SIP RTP TOS bits 184
>== Using SIP RTP CoS mark 5
>== Using SIP VRTP TOS bits 136
>== Using SIP VRTP CoS mark 4
>== Using UDPTL TOS bits 184
>== Using UDPTL CoS mark 5
> -- Called office-home-sip/170
>
>
> On the office asterisk:
>
> == Using SIP RTP TOS bits 184
>== Using SIP RTP CoS mark 5
>== Using SIP VRTP CoS mark 6
>== Using UDPTL TOS bits 184
>== Using UDPTL CoS mark 5
> -- Executing [...@default:1] Macro("SIP/xxx.yyy.zzz.aaa-0176",
> "stdexten,170,SIP/170") in new stack
> -- Executing [...@macro-stdexten:1]
> NoOp("SIP/xxx.yyy.zzz.aaa-0176", ""CallerID is: ""asterisk"
> ") in new stack
> -- Executing [...@macro-stdexten:2]
> Dial("SIP/xxx.yyy.zzz.aaa-0176", "SIP/170,18,rtT") in new stack
>== Using SIP RTP TOS bits 184
>== Using SIP RTP CoS mark 5
>== Using SIP VRTP CoS mark 6
>== Using UDPTL TOS bits 184
>== Using UDPTL CoS mark 5
> -- Called 170
>
> Why isn't the office asterisk picking up the callerid from the home
> asterisk?
>
> sean
>
>
Ping.
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Re: [asterisk-users] MATH
ok.
that worked
thanks!!
On Sun, Jan 31, 2010 at 10:50 AM, Tzafrir Cohen
wrote:
> On Sun, Jan 31, 2010 at 10:37:29AM -0500, Thomas Perron wrote:
>> hi
>> i don't claim to be a star at this but there must be some obvious part
>> missing;
>> my dial plan is below. out put from cli follows.
>>
>> exten => 3011,1,Answer()
>> exten => 3011,n,Set(TOTAL=0)
>> exten => 3011,n,Set(TOTAL=${Math(${TOTAL}+300,int)})
>> exten => 3011,n,WaitExten(3)
>> exten => 988,1,SayNumber(${TOTAL})
>>
>> [Jan 31 10:21:35] ERROR[1318]: pbx.c:2770 ast_func_read: Function Math
>> not registered
>
> Function names are CaSe SenSitive, and are normally ALL CAPS. You should
> use 'MATH' instead of 'Math'.
>
> /me is done shouting for today, hopefully.
>
> --
> Tzafrir Cohen
> icq#16849755 jabber:tzafrir.co...@xorcom.com
> +972-50-7952406 mailto:tzafrir.co...@xorcom.com
> http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
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> http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [asterisk-users] MATH
On Sun, Jan 31, 2010 at 10:37:29AM -0500, Thomas Perron wrote:
> hi
> i don't claim to be a star at this but there must be some obvious part
> missing;
> my dial plan is below. out put from cli follows.
>
> exten => 3011,1,Answer()
> exten => 3011,n,Set(TOTAL=0)
> exten => 3011,n,Set(TOTAL=${Math(${TOTAL}+300,int)})
> exten => 3011,n,WaitExten(3)
> exten => 988,1,SayNumber(${TOTAL})
>
> [Jan 31 10:21:35] ERROR[1318]: pbx.c:2770 ast_func_read: Function Math
> not registered
Function names are CaSe SenSitive, and are normally ALL CAPS. You should
use 'MATH' instead of 'Math'.
/me is done shouting for today, hopefully.
--
Tzafrir Cohen
icq#16849755 jabber:tzafrir.co...@xorcom.com
+972-50-7952406 mailto:tzafrir.co...@xorcom.com
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Re: [asterisk-users] MATH
hi
i don't claim to be a star at this but there must be some obvious part missing;
my dial plan is below. out put from cli follows.
exten => 3011,1,Answer()
exten => 3011,n,Set(TOTAL=0)
exten => 3011,n,Set(TOTAL=${Math(${TOTAL}+300,int)})
exten => 3011,n,WaitExten(3)
exten => 988,1,SayNumber(${TOTAL})
[Jan 31 10:21:35] ERROR[1318]: pbx.c:2770 ast_func_read: Function Math
not registered
-- Executing [3...@default:3] Set("SIP/64.85.162.137-c0132f50",
"TOTAL=") in new stack
-- Executing [3...@default:4]
WaitExten("SIP/64.85.162.137-c0132f50", "3") in new stack
[Jan 31 10:21:38] WARNING[1318]: pbx.c:7855 pbx_builtin_waitexten:
Timeout but no rule 't' in context 'default'
== Spawn extension (default, 3011, 4) exited non-zero on
'SIP/64.85.162.137-c0132f50'
localhost*CLI> Function Math not registered
No such command ' Function Math not registered' (type 'help Function
Math' for other possible commands)
2010/1/31 Håkon Nessjøen :
> You probably have to do a
>
> exten => s,1,n,Set(TOTAL=0)
>
> in the start of the call, to initialize the TOTAL variable
>
> On Sun, Jan 31, 2010 at 4:29 AM, Thomas Perron
> wrote:
>>
>> thanks for the response.
>> I tried to simplify and am now tuning the following, but it is not
>> responding with anything.
>> something wrong with timing?
>> here is what I have:
>>
>> exten => 1625,1,Answer()
>> exten => 1625,n,Set(TOTAL=${MATH(${TOTAL}+500,int)})
>> exten => 1625,n,WaitExten(3)
>> exten => 9625,1,Answer()
>> exten => 9625,n,SayNumber(${TOTAL})
>>
>>
>> output from the console
>>
>> [Jan 30 22:25:16] WARNING[22987]: func_math.c:194 math: '' is not a valid
>> number
>> -- Executing [1...@default:2] Set("SIP/64.85.162.137-c00d10e0",
>> "TOTAL=") in new stack
>> -- Executing [1...@default:3]
>> WaitExten("SIP/64.85.162.137-c00d10e0", "3") in new stack
>> [Jan 30 22:25:19] WARNING[22987]: pbx.c:7855 pbx_builtin_waitexten:
>> Timeout but no rule 't' in context 'default'
>> == Spawn extension (default, 1625, 3) exited non-zero on
>> 'SIP/64.85.162.137-c00d10e0'
>>
>>
>> 2010/1/30 Håkon Nessjøen :
>> > Try something like:
>> >
>> > exten => 1,1,WaitExten(3)
>> > exten => 1,1,Set(TOTAL=${MATH(${TOTAL}+500,int)})
>> > exten => 1,n,WaitExten(3)
>> > exten => 2,1,Set(TOTAL=${MATH(${TOTAL}+200,int)})
>> > exten => 2,n,WaitExten(3)
>> > exten => 3,1,Set(TOTAL=${MATH(${TOTAL}+300,int)})
>> > exten => 3,n,WaitExten(3)
>> > exten => 9,1,SayNumber(${TOTAL})
>> >
>> > Or something. Never used either math or saynumber before, but according
>> > to
>> > the documentation, something like this should work..
>> >
>> >
>> > On Sat, Jan 30, 2010 at 3:06 PM, Thomas Perron
>> > wrote:
>> >>
>> >> total up for current call.
>> >> then read back the number
>> >>
>> >>
>> >>
>> >> 2010/1/30 Håkon Nessjøen :
>> >> > For all calls combined, or for the current call?
>> >> >
>> >> > On Sat, Jan 30, 2010 at 2:48 PM, Thomas Perron
>> >> >
>> >> > wrote:
>> >> >>
>> >> >> I want to create a script for IVR that compiles responses,
>> >> >> aggregates
>> >> >> them to a total number.
>> >> >> Then, run an equation based on the result.
>> >> >>
>> >> >> Press 1 for X (X is a positive number 500)
>> >> >> Press 2 for Y (Y is a positive number 200)
>> >> >> Press 3 for Z (Z is a positive number 300)
>> >> >>
>> >> >> Press 20 to calculate the results
>> >> >> = 500+200+300 =1000
>> >> >> then,
>> >> >> exten => s,n,Read(NUMBER,,1000)
>> >> >> exten => s,n,SayDigits(${NUMBER})
>> >> >>
>> >> >> --
>> >> >>
>> >> >> _
>> >> >> -- 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
>> >> >
>> >>
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>> >> http://lists.digium.com/mailman/listinfo/asterisk-users
>> >
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>> >
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Re: [asterisk-users] Asterisk IPv6 update - we need an update
30 jan 2010 kl. 23.40 skrev Michiel van Baak: > On 14:29, Sat 30 Jan 10, Olle E. Johansson wrote: >> Friends, >> >> Before the Christmas holidays, I did send this letter and did not get a lot >> of response, but some. Since then, I've been able to get interest from a few >> parties that are willing to fund parts of this work, including Digium, the >> main sponsor of Asterisk. I will also apply for additional funding from a >> foundation here in Sweden and hope to get some more responses so that we can >> fund this project together. If anyone out there has interest or feedback >> regarding IPv6, Asterisk and VoIP, I'll be happy to get in contact. >> >> I've documented some of my thoughts on how to proceed, based on the work >> already done by Marc Blanchet (and of course work together with him) on my >> blog, http://www.voip-forum.com/asterisk/2010-01/voip-users-care-ipv6/ >> >> My hope is that we can get this done and integrated in Asterisk 1.8, but >> that requires some immediate attention from the community, as well as help >> with testing and feedback when we start rolling. Marcs code is already out >> there, so you can start testing NOW in your IPv6-enabled network. >> http://www.asteriskv6.org/ >> >> IPv6 is a boring topic, and if you do it right, no one will thank you for >> it. It just needs to be done. My work with IPv6 started the summer of 1995 >> and since then people have been shouting "We need to migrate now!". We've >> done that so long so that no one listens any more and now it's getting >> really critical. The IP numbering authorities, like ARIN and RIPE, have >> already outlined how they will have to change procedures for IPv4 >> assignments every six months from now, making it harder and harder to get >> addresses. For VoIP - sip trunks, calling each other across the Internet, >> it's critical to have public IP addresses unless you want to stay with your >> lovely Telco on the other end of the copper cables. >> >> Personally, I'm not sure how to design software for this migration properly. >> In order to educate myself and collegues that develop and build SIP >> solutions, I'm going to organize an event this spring which combines testing >> and training. I do hope that the Asterisk community will join me and support >> the developer team in our efforts to make Asterisk - the leading Open Source >> PBX - running perfectly well on both IPv4 and IPv6 networks. It needs to >> be done, we will get it done. And no one will thank us for it, since >> everyone just expects Asterisk to work as we have done for the last 10 >> years... >> >> With IPv6 greetings! > > Says he, who sent this mail over a pure IPv4 network. > Not a single IPv6 hop in the path. tsk tsk tsk. EXACTLY! It's embarrasing. Everytime I turn on IPv6 in my local network, some application break. This is why I want to start fixing this! > > Like I said before, I'm really interested in this, and have a couple of > ipv6 boxen with asterisk on it. I also have some credits with sixxs.net > so I can provide at least 1 subnet for testing. We'll make sure you get involved, don't worry. ;-) /O -- _ -- 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 IPv6 update - we need an update
On 14:29, Sat 30 Jan 10, Olle E. Johansson wrote: > Friends, > > Before the Christmas holidays, I did send this letter and did not get a lot > of response, but some. Since then, I've been able to get interest from a few > parties that are willing to fund parts of this work, including Digium, the > main sponsor of Asterisk. I will also apply for additional funding from a > foundation here in Sweden and hope to get some more responses so that we can > fund this project together. If anyone out there has interest or feedback > regarding IPv6, Asterisk and VoIP, I'll be happy to get in contact. > > I've documented some of my thoughts on how to proceed, based on the work > already done by Marc Blanchet (and of course work together with him) on my > blog, http://www.voip-forum.com/asterisk/2010-01/voip-users-care-ipv6/ > > My hope is that we can get this done and integrated in Asterisk 1.8, but that > requires some immediate attention from the community, as well as help with > testing and feedback when we start rolling. Marcs code is already out there, > so you can start testing NOW in your IPv6-enabled network. > http://www.asteriskv6.org/ > > IPv6 is a boring topic, and if you do it right, no one will thank you for it. > It just needs to be done. My work with IPv6 started the summer of 1995 and > since then people have been shouting "We need to migrate now!". We've done > that so long so that no one listens any more and now it's getting really > critical. The IP numbering authorities, like ARIN and RIPE, have already > outlined how they will have to change procedures for IPv4 assignments every > six months from now, making it harder and harder to get addresses. For VoIP - > sip trunks, calling each other across the Internet, it's critical to have > public IP addresses unless you want to stay with your lovely Telco on the > other end of the copper cables. > > Personally, I'm not sure how to design software for this migration properly. > In order to educate myself and collegues that develop and build SIP > solutions, I'm going to organize an event this spring which combines testing > and training. I do hope that the Asterisk community will join me and support > the developer team in our efforts to make Asterisk - the leading Open Source > PBX - running perfectly well on both IPv4 and IPv6 networks. It needs to be > done, we will get it done. And no one will thank us for it, since everyone > just expects Asterisk to work as we have done for the last 10 years... > > With IPv6 greetings! Says he, who sent this mail over a pure IPv4 network. Not a single IPv6 hop in the path. tsk tsk tsk. Like I said before, I'm really interested in this, and have a couple of ipv6 boxen with asterisk on it. I also have some credits with sixxs.net so I can provide at least 1 subnet for testing. Money on the other hand is a totally different issue. Cant provide that. sorry. > > /Olle > > > Vidarebefordrat brev: > > > Fr?n: "Olle E. Johansson" > > Datum: 17 december 2009 09.39.40 CET > > Till: Asterisk Non-Commercial Discussion Users Mailing List - > > > > ?mne: [asterisk-users] Asterisk IPv6 update - we need an update > > Svara till: Asterisk Users Mailing List - Non-Commercial Discussion > > > > > > Friends, > > > > At the first Astricon I was very happy to see Marc Blanchet as one of the > > attendees. I knew he was one of the IPv6 gurus and I wanted someone to show > > some interest in Asterisk and IPv6. > > > > Well, he did not only get interested in it, but started coding on it. The > > results have been available for quite some time at > > http://www.asteriskv6.org/ and Marc has tested it at several SIPits for > > interoperability. > > > > This patch is very large and affects large areas of Asterisk. In order to > > support IPv6, we need to update the way we interact with sockets, with DNS, > > with URI's. The SIP channel needs to handle multiple UDP as well as TCP > > sockets in both protocols. The ACL's we use for all VoIP protocols and > > manager needs support for IPv6. And much more. > > > > Marc hasn't been able to spend time to keep it up to date with the > > everchanging trunk. > > > > I feel we need to move this forward and try to divide the large patch into > > smaller pieces that can be reviewed separately by the developer team and > > be merged gradually. First, Marcs branch needs a serious overhaul to get up > > to date with trunk. In order to work on this, Marc and I needs funding. > > > > I have a few interested parties, but need more interested parties that can > > commit to funding during the first half of 2010 for this project. It's not > > a small task, the current estimate is at least one month's work for each of > > us for updating, cutting it up, merging, going through the review process, > > testing and finalizing with new tests at SIPit or a similar event. > > > > If your organization is interested, please let me know off list
Re: [asterisk-users] Unable to create channel of type 'DAHDI'(cause 0- Unknown)
On Fri, Jan 29, 2010 at 05:48:53PM -0500, sean darcy wrote: > listu...@spamomania.co.uk wrote: > > On Thu, 2010-01-28 at 23:11 -0600, Karl Fife wrote: > >> Appears completely resolved! > >> No more home-spun patches! > >> Thanks! > >> -K > >> > > It's *not* fixed here: > > DAHDI Version: 2.2.1 Echo Canceller: MG2 > > > > But as is depressingly the 'norm' for Asterisk it comes back to bitching > > about hardware 'buy an expensive Digium echo machine instead of a cheap > > one' rather than the fact that the core of Asterisk is rotten, buggy and > > the fix usually comes in the form of a developer arguing that it's > > somebody else's issue. > > > > Really - if Asterisk is 'The future of telephony' I can only assume that > > statement comes from the late 1800's. If you like echo, flaky > > connections, intermittent service and partially working DTMF coupled > > with a hefty hardware price tag then hey ho - Asterisk is your man > > Nice try, be great when it's finished. > > > > > > Sigh. > > OK you don't like asterisk - sorry. Obviously some other software works > better for you. I'm glad. > > For at least some of us, asterisk works extremely well in demanding > environments. But not perfectly. So the collegial help from the mailing > list and bug spotting is quite important. > > Sorry you don't want to participate. FWIW, I also consider Asterisk's behaviour here buggy. http://svn.debian.org/viewsvn/pkg-voip/asterisk/trunk/debian/patches/dahdi-fxsks-hookstate?view=markup -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Astribank problem
On Sat, Jan 30, 2010 at 03:57:30AM +, frangky robert wrote: > > H all... > > I have an Astribank (8FXS/16FXO), IBM X3200 M2, Asterisk-1.6.2.1, > dahdi-linux-complete-2.2.1, libpri-1.4.10.2, centos-5.4 final. > My problem is, every time i unplug the astribank power supply, and > reconnect it, astribank cannot work again (lsusb result is 11x0)... When you plug the Astribank, the firmware should get loaded by a script called from udev. What messages do you see in /var/log/messages following that? > but, after reinstall the asterisk and dahdi, astribank will detected > (lsusb result is 11x2)... -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MATH
You probably have to do a
exten => s,1,n,Set(TOTAL=0)
in the start of the call, to initialize the TOTAL variable
On Sun, Jan 31, 2010 at 4:29 AM, Thomas Perron wrote:
> thanks for the response.
> I tried to simplify and am now tuning the following, but it is not
> responding with anything.
> something wrong with timing?
> here is what I have:
>
> exten => 1625,1,Answer()
> exten => 1625,n,Set(TOTAL=${MATH(${TOTAL}+500,int)})
> exten => 1625,n,WaitExten(3)
> exten => 9625,1,Answer()
> exten => 9625,n,SayNumber(${TOTAL})
>
>
> output from the console
>
> [Jan 30 22:25:16] WARNING[22987]: func_math.c:194 math: '' is not a valid
> number
>-- Executing [1...@default:2] Set("SIP/64.85.162.137-c00d10e0",
> "TOTAL=") in new stack
>-- Executing [1...@default:3]
> WaitExten("SIP/64.85.162.137-c00d10e0", "3") in new stack
> [Jan 30 22:25:19] WARNING[22987]: pbx.c:7855 pbx_builtin_waitexten:
> Timeout but no rule 't' in context 'default'
> == Spawn extension (default, 1625, 3) exited non-zero on
> 'SIP/64.85.162.137-c00d10e0'
>
>
> 2010/1/30 Håkon Nessjøen :
> > Try something like:
> >
> > exten => 1,1,WaitExten(3)
> > exten => 1,1,Set(TOTAL=${MATH(${TOTAL}+500,int)})
> > exten => 1,n,WaitExten(3)
> > exten => 2,1,Set(TOTAL=${MATH(${TOTAL}+200,int)})
> > exten => 2,n,WaitExten(3)
> > exten => 3,1,Set(TOTAL=${MATH(${TOTAL}+300,int)})
> > exten => 3,n,WaitExten(3)
> > exten => 9,1,SayNumber(${TOTAL})
> >
> > Or something. Never used either math or saynumber before, but according
> to
> > the documentation, something like this should work..
> >
> >
> > On Sat, Jan 30, 2010 at 3:06 PM, Thomas Perron
> > wrote:
> >>
> >> total up for current call.
> >> then read back the number
> >>
> >>
> >>
> >> 2010/1/30 Håkon Nessjøen :
> >> > For all calls combined, or for the current call?
> >> >
> >> > On Sat, Jan 30, 2010 at 2:48 PM, Thomas Perron <
> thomas.per...@gmail.com>
> >> > wrote:
> >> >>
> >> >> I want to create a script for IVR that compiles responses, aggregates
> >> >> them to a total number.
> >> >> Then, run an equation based on the result.
> >> >>
> >> >> Press 1 for X (X is a positive number 500)
> >> >> Press 2 for Y (Y is a positive number 200)
> >> >> Press 3 for Z (Z is a positive number 300)
> >> >>
> >> >> Press 20 to calculate the results
> >> >> = 500+200+300 =1000
> >> >> then,
> >> >> exten => s,n,Read(NUMBER,,1000)
> >> >> exten => s,n,SayDigits(${NUMBER})
> >> >>
> >> >> --
> >> >> _
> >> >> -- 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
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> >
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> >
>
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>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
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