[Asterisk-Users] Transient SIP Registration Issues
Hey Everyone -
I am having a problem that is keeping me awake at night.ok, so maybe
not keeping me awake, but it is frustrating. :-)
I am running Asterisk 1.0.7 on Gentoo (2.6.10-gentoo-r6) on an Intel
700Mhz box with 512MB of RAM.
The system is very light, with maybe 35 SIP and IAX connections. I am
using NuFone and Konfer for dialtone with no traditional TDM cards
installed at all. Overall system load is around .4 or less most of the
time.
Overall - a very simple configuration.
I am using (mostly) the Linksys PAP2-NA units for deployment. I
preconfigure the units, then ship them out to the people that need them.
I also have several of the Digium IAXy units in use.
The problem I am starting to see is that a person's extension will work
great, and then I will start to see failed registrations for their unit
over and over again. When this happens, the units fall offline. Then the
unit will magically reregister and start to work again.
I had assumed (initially) that it was a bad unit, so I replaced it, but
then it started to happen to other units as well.
When registered, the units in question have ping time under 50 to 60 ms,
and no latency associated with them. Packet loss is extremely minimal or
none at all.
Here is one example - I have included the relevant portions of my
sip.conf and extensions.conf:
sip.conf
[1028]
type=friend
username=GSynn
secret=
qualify=500
host=dynamic
fromuser=GSynn
dtmfmode=rfc2833
nat=yes
canreinvite=no
disallow=all
allow=g729
callerid=Gary Synn 3050
context=secure
extension.conf
[gary_synn]
exten = 3050,1,Macro(stdexten,SIP/1028)
[macro-stdexten]
;; ARG1 = Phone ID to dial
exten = s,1,NoOp(${CALLERID}) ; Grab Caller ID Info
exten = s,2,Playback(${VMDIR}/${MACRO_EXTEN}/greet); Grab the
dialed extension and play a greeting
exten = s,3,Dial(${ARG1},15,rtm)
exten = s,4,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Voicemail(u${MACRO_EXTEN})
exten = s-CHANUNAVAIL,1,Voicemail(u${MACRO_EXTEN}) ;if chan
unavail (sip phone not regisitered?)
exten = s-CONGESTION,1,Voicemail(u${MACRO_EXTEN}) ;if chan
congested
exten = s-BUSY,1,Voicemail(b${MACRO_EXTEN})
exten = s-.,1,Goto(s-NOANSWER,1)
exten = a,1,VoicemailMain(${MACRO_EXTEN})
[secure]
include = pstn_outbound
include = system_extensions
Here are the errors that I see on the colsole -
Apr 4 17:35:38 NOTICE[27276]: chan_sip.c:7681 handle_request:
Registration from 'GSynn sip:[EMAIL PROTECTED]' failed for
'5.63.198.220'
Apr 4 17:35:42 NOTICE[27276]: chan_sip.c:7681 handle_request:
Registration from 'GSynn sip:[EMAIL PROTECTED]' failed for
'5.63.198.220'
Apr 4 17:35:46 NOTICE[27276]: chan_sip.c:7681 handle_request: Registration
from 'GSynn sip:[EMAIL PROTECTED]' failed for '5.63.198.220'
Apr 4 17:35:50 NOTICE[27276]: chan_sip.c:7681 handle_request: Registration
from 'DMadore sip:[EMAIL PROTECTED]' failed for '69.17.136.238'
Apr 4 17:35:50 NOTICE[27276]: chan_sip.c:7681 handle_request: Registration
from 'GSynn sip:[EMAIL PROTECTED]' failed for '5.63.198.220'
Apr 4 17:35:54 NOTICE[27276]: chan_sip.c:7681 handle_request: Registration
from 'GSynn sip:[EMAIL PROTECTED]' failed for '5.63.198.220'
Apr 4 17:35:58 NOTICE[27276]: chan_sip.c:7681 handle_request: Registration
from 'GSynn sip:[EMAIL PROTECTED]' failed for '5.63.198.220'
Apr 4 17:36:02 NOTICE[27276]: chan_sip.c:7681 handle_request: Registration
from 'GSynn sip:[EMAIL PROTECTED]' failed for '5.63.198.220'
Apr 4 17:36:33 NOTICE[27276]: chan_sip.c:7681 handle_request:
Registration from 'GSynn sip:[EMAIL PROTECTED]' failed for
'5.63.198.220'
Apr 4 17:36:34 NOTICE[27276]: chan_sip.c:7681 handle_request: Registration
from 'GSynn sip:[EMAIL PROTECTED]' failed for '5.63.198.220'
Apr 4 17:36:37 NOTICE[27276]: chan_sip.c:7681 handle_request: Registration
from 'GSynn sip:[EMAIL PROTECTED]' failed for '5.63.198.220'
Apr 4 17:36:40 NOTICE[27276]: chan_sip.c:7681 handle_request: Registration
from 'GSynn sip:[EMAIL PROTECTED]' failed for '5.63.198.220'
Apr 4 17:36:44 NOTICE[27276]: chan_sip.c:7681 handle_request:
Registration from 'GSynn sip:[EMAIL PROTECTED]' failed for
'5.63.198.220'
Apr 4 17:36:48 NOTICE[27276]: chan_sip.c:7681 handle_request: Registration
from 'GSynn sip:[EMAIL PROTECTED]' failed for '5.63.198.220'
Apr 4 17:36:52 NOTICE[27276]: chan_sip.c:7681 handle_request: Registration
from 'GSynn sip:[EMAIL PROTECTED]' failed for '5.63.198.220'
Any ideas would be greatly appreciated.
Thanks !!
**
Richard J. Sears
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Re: [Asterisk-Users] bandwidth
We use ulaw where we can and g729 where necessary. I think it is like 8k for g729. On Mon, 04 Apr 2005 19:07:24 -0500 Bernie [EMAIL PROTECTED] wrote: can that number be reduced? I'm looking at a system that would be deployed to remote offices over fairly limited bandwidth links and need to find a way of balancing quality vs. bandwidth constraints. B William Boehlke wrote: The simple answer is 64KB. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bernie Sent: Monday, April 04, 2005 4:40 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] bandwidth how much bandwith is used to go between a phone set and the asterisk server when a call is in progress? Just trying to plan out a system and need some figures to plan on bandwidth allocation. B ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ** Richard J. Sears Vice President American Internet Services [EMAIL PROTECTED] http://www.adnc.com 858.576.4272 - Phone 858.427.2401 - Fax INOC-DBA - 6130 I fly because it releases my mind from the tyranny of petty things . . Work like you don't need the money, love like you've never been hurt and dance like you do when nobody's watching. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] peering
Looks like you do not have the context set correctly in your iax.conf on both sides. Make sure that it exists and it is going to do what you want it to do. On Fri, 25 Mar 2005 18:57:06 +1000 AS [EMAIL PROTECTED] wrote: Our main asterisk box peers with that of a customer. We are trying to assign DID's to their extensions but get this error. What are we doing wrong? Client side Mar 25 18:49:47 NOTICE[1369]: chan_iax2.c:6545 socket_read: Rejected connect attempt from 203.xxx.xxx.16, who was trying to reach 's@' Our side Mar 25 18:56:15 WARNING[705]: chan_iax2.c:5546 socket_read: Call rejected by 203.xxx.xxx.17: No authority found -- Hungup 'IAX2/username/23' == No one is available to answer at this time ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ** Richard J. Sears Vice President American Internet Services [EMAIL PROTECTED] http://www.adnc.com 858.576.4272 - Phone 858.427.2401 - Fax INOC-DBA - 6130 I fly because it releases my mind from the tyranny of petty things . . Work like you don't need the money, love like you've never been hurt and dance like you do when nobody's watching. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Emailed voicemail
Yes Andy - that was my mistake. I have my system hacked up to do some other things. It should be: 1234 = 1234,Bob Jones,[EMAIL PROTECTED] do your mail logs have any errors at all in them in regards to mail bouncing or anything like that..? Do you have your servermail settings configured in voicemail.conf and did you (maybe) compile asterisk to use asterisk_vm mysql db instead of the voicemail.conf..? On Fri, 25 Mar 2005 04:55:48 -0500 Andy Stewart [EMAIL PROTECTED] wrote: Richard, Yep, got that config'd in there: 1001 = 1001,Andy Stewart,[EMAIL PROTECTED] 1002 = 1002,Lorri Barnett,[EMAIL PROTECTED] 1003 = 1003,Andy Stewart - Home,[EMAIL PROTECTED] 1004 = 1004,Andy Stewart - HardPhone,[EMAIL PROTECTED] 1005 = 1005,Lorri Barnett - HardPhone,[EMAIL PROTECTED] Or it this maybe the problem? Your example is ext = ext,emailMine above (and the example in voicemail.conf) is ext = ext,name,email ?? Thanx A From: Richard J. Sears [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Emailed voicemail To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=US-ASCII Hi Andy, did you configure voicemail.conf with the users e-mail address...? 1234 = 1234,[EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ** Richard J. Sears Vice President American Internet Services [EMAIL PROTECTED] http://www.adnc.com 858.576.4272 - Phone 858.427.2401 - Fax INOC-DBA - 6130 I fly because it releases my mind from the tyranny of petty things . . Work like you don't need the money, love like you've never been hurt and dance like you do when nobody's watching. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Emailed voicemail
Hi Andy, did you configure voicemail.conf with the users e-mail address...? 1234 = 1234,[EMAIL PROTECTED] On Thu, 24 Mar 2005 22:02:22 -0500 Andy Stewart [EMAIL PROTECTED] wrote: Have Asterisk us at running fine, but have run into a small snag. It's not emailing the voicemails to the users. I have attach=yes set, sendmail is configured and works from from the commandline (sent an email to myself). Unless I'm wrong, or missing something, asterisk is configured by default to send an email when a users receives a voicemail, correct? Thanx A ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ** Richard J. Sears Vice President American Internet Services [EMAIL PROTECTED] http://www.adnc.com 858.576.4272 - Phone 858.427.2401 - Fax INOC-DBA - 6130 I fly because it releases my mind from the tyranny of petty things . . Work like you don't need the money, love like you've never been hurt and dance like you do when nobody's watching. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] sip show peers weirdness
Hey Everyone, This is not an operational issue, and to my knowledge only effects the look of the command, but when I issue a sip reload then a sip show peers I see all of the actual usernames I have assigned in my sip.conf. However, five minutes later I reissue the sip show peers and all of the usernames have disappeared and are replaced by the sip ID. The only way to get them back is to do a sip reload again. Also a sip show users shows only the SIP id, no usernames at all. Any one have any clue on this one..? I am running 1.0.6 on Gentoo. Thanks ** Richard J. Sears Vice President American Internet Services [EMAIL PROTECTED] http://www.adnc.com 858.576.4272 - Phone 858.427.2401 - Fax INOC-DBA - 6130 I fly because it releases my mind from the tyranny of petty things . . Work like you don't need the money, love like you've never been hurt and dance like you do when nobody's watching. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: PC sound hardware for voice recording
Same here - I record all my prompts either over a regular phone (Cisco
7960) or using the Plantronics DSP 500 and the Record application.
exten = 405,1,AGI(set-timestamp.agi)
exten = 405,2,Playback(rjs-voice-prompt-recorder) ; Play some instructions
exten = 405,3,Wait(2)
exten = 405,4,Record(/tmp/asterisk-recording-${timestamp}:gsm)
exten = 405,5,Wait(2)
exten = 405,6,Playback(/tmp/asterisk-recording-${timestamp})
exten = 405,7,Wait(2)
exten = 405,8,Hangup
Then I move the file from /tmp to whereever I need it.
On Thu, 17 Mar 2005 11:14:01 -0700
Wiley Siler [EMAIL PROTECTED] wrote:
I recorded my last set of prompts over my Plantronics DSP 500 USB
Headset. I have also used a Logitech USB Headset. These and similar are
easiest to use along with X-lite or similar softphone. I used the
suggested method of dialing an extension on the PBX and letting Asterisk
record for me direct via the phone. The quality is just as good as
using a dedicated soundcard and mic IMHO. At $50-75 for the USB
headset, it is a good cheap investment.
Thanks,
Wiley
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of snacktime
Sent: Thursday, March 17, 2005 11:00 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] OT: PC sound hardware for voice recording
What would be a minimum sound card/microphone combo for good voice
quality recording on a budget? This would be for * voice prompts.
Would a soundblaster live and a good mic do the job?
Chris
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Richard J. Sears
Vice President
American Internet Services
[EMAIL PROTECTED]
http://www.adnc.com
858.576.4272 - Phone
858.427.2401 - Fax
INOC-DBA - 6130
I fly because it releases my mind
from the tyranny of petty things . .
Work like you don't need the money, love like you've
never been hurt and dance like you do when nobody's
watching.
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[Asterisk-Users] NuFone and CallerID
Hey Everyone, I am using NuFone for 866 inbound service and I am trying to figure out the callerid part of it. Any call into my * system just shows Toll Free Call and will not give me the calling party's caller ID info. Is this just something I have to live with using NuFOne, or did I miss some type of config in * that will grab the callerID other than the inbound 866 number...? Thanks !! ** Richard J. Sears Vice President American Internet Services [EMAIL PROTECTED] http://www.adnc.com 858.576.4272 - Phone 858.427.2401 - Fax INOC-DBA - 6130 I fly because it releases my mind from the tyranny of petty things . . Work like you don't need the money, love like you've never been hurt and dance like you do when nobody's watching. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] NuFone and CallerID
So how do I capture the caller number..? All I see is the 866 number that NuFone has assigned to me. Thanks !! On Wed, 16 Mar 2005 15:40:41 -0800 Scott Laird [EMAIL PROTECTED] wrote: On Mar 16, 2005, at 3:31 PM, Richard J. Sears wrote: Hey Everyone, I am using NuFone for 866 inbound service and I am trying to figure out the callerid part of it. Any call into my * system just shows Toll Free Call and will not give me the calling party's caller ID info. Is this just something I have to live with using NuFOne, or did I miss some type of config in * that will grab the callerID other than the inbound 866 number...? You should see the calling number as well, but NuFone doesn't provide caller name for 800 numbers. The last time this came up, it looked like most (if not all) VoIP 800-number providers were in the same boat. Scott ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ** Richard J. Sears Vice President American Internet Services [EMAIL PROTECTED] http://www.adnc.com 858.576.4272 - Phone 858.427.2401 - Fax INOC-DBA - 6130 I fly because it releases my mind from the tyranny of petty things . . Work like you don't need the money, love like you've never been hurt and dance like you do when nobody's watching. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] NuFone and CallerID
Hi Scott -
You are correct - this is in the CDR:
2005-03-16 16:38:32 IAX2/[EMAIL PROTECTED]:45 7607534720 Toll-Free Call
7607534720 Dial SIP/1022|15|rtm 2060 ANSWERED 00:28
So it looks like I am getting the caller ID to the phone, but all I am
seeing on any of the phones I have tested is the first part Toll-Free
Call -
It would appear that the phones that I am using do not have enough
screen space to show the number as well.
Thanks much for the help.
On Wed, 16 Mar 2005 15:55:51 -0800
Scott Laird [EMAIL PROTECTED] wrote:
On Mar 16, 2005, at 3:49 PM, Richard J. Sears wrote:
So how do I capture the caller number..? All I see is the 866 number
that NuFone has assigned to me.
What's in ${CALLERIDNUM}? It should also show up in the CDR. For me,
it works exactly like caller ID information generated by my X100P,
except for the missing name data and the usual ANI issues.
Scott
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Richard J. Sears
Vice President
American Internet Services
[EMAIL PROTECTED]
http://www.adnc.com
858.576.4272 - Phone
858.427.2401 - Fax
INOC-DBA - 6130
I fly because it releases my mind
from the tyranny of petty things . .
Work like you don't need the money, love like you've
never been hurt and dance like you do when nobody's
watching.
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Re: [Asterisk-Users] Asterisk Brochure
Digium Has a pretty nice one on their site: http://www.digium.com/downloads/marketing/asterisk.pdf On Fri, 4 Mar 2005 22:57:05 -0600 Anton Krall [EMAIL PROTECTED] wrote: Guys. Anybody has developed and asterisk brochure for commercial purposes (consultant, etc) that I might be able to take a look at? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ** Richard J. Sears Vice President American Internet Services [EMAIL PROTECTED] http://www.adnc.com 858.576.4272 - Phone 858.427.2401 - Fax INOC-DBA - 6130 I fly because it releases my mind from the tyranny of petty things . . Work like you don't need the money, love like you've never been hurt and dance like you do when nobody's watching. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Forwarding to Cell Phone, Pager, etc
Hi Neel -
We did it by dialing several devices at once, then calling the cell and
then pulling the call back if no one answered the cell to allow the
local vm to handle vm functions. Here is how we did it:
[macro-stdexten_cell]
; ARG1 = Greeting
; ARG2 = Extension(s) to dial
; ARG3 = Cell Phone Number to dial
;
exten = s,1,Playback(${ARG1})
exten = s,2,Dial(${ARG2},15,rtm)
exten = s,3,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Dial(IAX2/[EMAIL PROTECTED]/${ARG3},15,rtm)
exten = s-NOANSWER,2,Voicemail(u${MACRO_EXTEN})
exten = s-CHANUNAVAIL,1,Voicemail(b${MACRO_EXTEN}) ;if chan
unavail (sip phone not regisitered?)
;exten = s-NOANSWER,3,Goto(default,s,1)
exten = s-BUSY,1,Voicemail(b${MACRO_EXTEN})
;exten = s-BUSY,2,Goto(default,s,1)
exten = s-.,1,Goto(s-NOANSWER,1)
exten = a,1,VoicemailMain(${MACRO_EXTEN})
Then here is the extension entry that made it happen:
exten =
${RJSWORK},1,Macro(stdexten_cell,sears_welcome,SIP/${RJSWORK}SIP/${RJSDESK}SIP/${RJSLAPTOP}SIP/${RJSHOME},${RJSCELL})
All the variables are just the extensions numbers.
Hope this helps
On Wed, 2 Mar 2005 11:56:14 -0800
Nitesh Divecha [EMAIL PROTECTED] wrote:
Hello all,
Was just wondering if Asterisk can do Call forwarding to cell phones,
pagers, home phone, etc.
For example, if exten 202 is away, he will set his call blasting priority
like first ring the exten for 10 sec if not answered then ring cell number
for 10 sec again if not answered, then ring home and etc.
Like a call blasting priority...
Any help would be appreciated.
Neel
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Richard J. Sears
Vice President
American Internet Services
[EMAIL PROTECTED]
http://www.adnc.com
858.576.4272 - Phone
858.427.2401 - Fax
INOC-DBA - 6130
I fly because it releases my mind
from the tyranny of petty things . .
Work like you don't need the money, love like you've
never been hurt and dance like you do when nobody's
watching.
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Re: [Asterisk-Users] Delay after entering digits with IVR
You were correct Steven - I was picking up the extensions from an
include after a jump !!
Lesson Learned - thanks everyone.
On Thu, 24 Feb 2005 20:18:22 -0600
Steven Critchfield [EMAIL PROTECTED] wrote:
On Thu, 2005-02-24 at 15:49 -0800, Richard J. Sears wrote:
I have a [start] context that all my inbound and '0' calls are routed
into.
Because of the way I want to set my system up, I want to prompt the user
to enter a 1 if they know the extension, or a 2 for a directory and
nothing else.
It works, however there is a 5 to 10 second delay after enter the 1 or 2
before the system responds.
I have read over the wiki on how asterisk handles digit inputs, but
cannot seem to isolate the problem. No other extension beginning with
(or even including) a '1' or a '2'.
Is this just how the system operates, or am I missing something..?
If That is the entirety of your start context, then it shouldn't be
doing any delay between detection and beginning action.
So my question is, is it possible that the delay is actually in the next
step such as the goto that jumps out to a different extension and
context or in the starting of the directory app.
Here is the [start] in my extensions.conf :
[start]
; If someone dials the Operator, just start them here.
exten = 0,1,Goto(s,1)
exten = s,1,Wait,1 ; Wait a second, just for fun
exten = s,2,Answer ; Answer the line
exten = s,3,SetMusicOnHold,default
exten = s,4,ResponseTimeout,5 ; Set Response Timeout
; Is is Morning, Afternoon or Evening ?
; Lets play a differnet greeting for each time period.
exten = s,5,AGI(openclose.agi)
exten = s,6,GotoIF($[${STATUS} = morning]?10)
exten = s,7,GotoIF($[${STATUS} = afternoon]?12)
exten = s,8,GotoIF($[${STATUS} = evening]?14)
extex = s,9,Goto(s,6)
; The various Greetings based on Time of Day
exten = s,10,Background(rjs-morning-welcome)
exten = s,11,Goto(s,15)
exten = s,12,Background(rjs-afternoon-welcome)
exten = s,13,Goto(s,15)
exten = s,14,Background(rjs-evening-welcome)
; The Voice Menu
exten = s,15,Background(rjs-if-you-know-the-extension)
exten = s,16,Wait,1
exten = s,17,BackGround(to-dial-by-name-press) ; Play some instructions
exten = s,18,BackGround(digits/2) ; Play some instructions
; A timeout and invalid extension rule
;
exten = t,1,Goto(s,15) ; If they take too long, give up
exten = i,1,Playback(invalid) ; That's not valid, try again
; If they know the extension, send them on.
exten = 1,1,Goto(extension_is_known,s,1)
; Allow users the ability to get Directory listing (user must be in
voicemail.conf)
exten = 2,1,Directory,default|internal_extensions
--
Steven Critchfield [EMAIL PROTECTED]
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American Internet Services
[EMAIL PROTECTED]
http://www.adnc.com
858.576.4272 - Phone
858.427.2401 - Fax
INOC-DBA - 6130
I fly because it releases my mind
from the tyranny of petty things . .
Work like you don't need the money, love like you've
never been hurt and dance like you do when nobody's
watching.
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[Asterisk-Users] WebVMail Woirks but No Audio
Hi Everyone - I have webvmail up and running, I can see the messages, forward them, pretty much everything but listen to them. Here is what I see in my logs: 192.168.0.1 - - [25/Feb/2005:08:15:40 -0800] GET /vmail/vmail.cgi?action=audiofolder=INBOXmailbox=2377context=default password=12msgid=format=gsmdontcasheme=4624.gsm HTTP/1.1 200 9438 - contype But the box at the bottom shows up as a broken link. Any ideas...? Thanks ** Richard J. Sears Vice President American Internet Services [EMAIL PROTECTED] http://www.adnc.com 858.576.4272 - Phone 858.427.2401 - Fax INOC-DBA - 6130 I fly because it releases my mind from the tyranny of petty things . . Work like you don't need the money, love like you've never been hurt and dance like you do when nobody's watching. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] WebVMail Woirks but No Audio
Hi Assaf - Already did that - the audio app location shows as a broken link on the page and plays nothing. On Fri, 25 Feb 2005 14:06:41 -0500 Assaf Benharoosh [EMAIL PROTECTED] wrote: I had this issue- it's security on the files. I put a cron job that do /bin/chmod 777 /var/spool/asterisk/voicemail/default -R evey 1 minute, but there may be a cleaner solution. Assaf Benharoosh MCP, MCSA, MCSE [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Richard J. Sears Sent: Friday, February 25, 2005 11:20 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] WebVMail Woirks but No Audio Hi Everyone - I have webvmail up and running, I can see the messages, forward them, pretty much everything but listen to them. Here is what I see in my logs: 192.168.0.1 - - [25/Feb/2005:08:15:40 -0800] GET /vmail/vmail.cgi?action=audiofolder=INBOXmailbox=2377context=default password=12msgid=format=gsmdontcasheme=4624.gsm HTTP/1.1 200 9438 - contype But the box at the bottom shows up as a broken link. Any ideas...? Thanks ** Richard J. Sears Vice President American Internet Services [EMAIL PROTECTED] http://www.adnc.com 858.576.4272 - Phone 858.427.2401 - Fax INOC-DBA - 6130 I fly because it releases my mind from the tyranny of petty things . . Work like you don't need the money, love like you've never been hurt and dance like you do when nobody's watching. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ** Richard J. Sears Vice President American Internet Services [EMAIL PROTECTED] http://www.adnc.com 858.576.4272 - Phone 858.427.2401 - Fax INOC-DBA - 6130 I fly because it releases my mind from the tyranny of petty things . . Work like you don't need the money, love like you've never been hurt and dance like you do when nobody's watching. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] I need to dial multiple numbers concurently but with delays.
I did the same thing, only I ring 3 phones at once and then my cell,
dropping back to my system VM if I don't answer my cell.
Here is how I did it:
I applied some variables to make it easer:
RJSVMEXT=2377
RJSWORK=2377
RJSDESK=2378
RJSLAPTOP=2379
RJSHOME=2380
RJSCELL=18585551212
I created the macro to do the dialing:
[macro-stdexten_cell]
; ARG1 = Welcome Message to play
; ARG2 = Phones to Ring
; ARG3 = Cell Phone Number
; Ring any phone passed to the macro for 15 seconds with
; music background, then ring my cell phone for 15 seconds
; then back to system vmail if no answer.
exten = s,1,Playback(${ARG1})
exten = s,2,Dial(${ARG2},15,rtm)
exten = s,3,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Dial(IAX2/[EMAIL PROTECTED]/${ARG3},15,rtm)
exten = s-NOANSWER,2,Voicemail(u${MACRO_EXTEN})
exten = s-BUSY,1,Voicemail(b${MACRO_EXTEN})
exten = s-.,1,Goto(s-NOANSWER,1)
exten = a,1,VoicemailMain(${MACRO_EXTEN})
Then I created the extension entry:
exten =
${RJSWORK},1,Macro(stdexten_cell,sears_welcome,SIP/${RJSWORK}SIP/${RJSDESK}SIP/${RJSLAPTOP}SIP/${RJSHOME},${RJSCELL})
Hope this helps.
On Sat, 19 Feb 2005 18:59:09 +0200
[EMAIL PROTECTED] wrote:
I have let's say a reception that is comprised of 2 zap extensions and a
mobile phone to dial using ISDN through Capi.
I want to have a delay before starting dialing the mobile phone so that it
rings only when the call has been unanswered for say 25 seconds.
I tried to use Capi/210699:ww693555 but without any success.
There is any way to do it or the code has to be modified ?
Thanks
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**
Richard J. Sears
Vice President
American Internet Services
[EMAIL PROTECTED]
http://www.adnc.com
858.576.4272 - Phone
858.427.2401 - Fax
INOC-DBA - 6130
I fly because it releases my mind
from the tyranny of petty things . .
Work like you don't need the money, love like you've
never been hurt and dance like you do when nobody's
watching.
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[Asterisk-Users] Delay after entering digits with IVR
I have a [start] context that all my inbound and '0' calls are routed
into.
Because of the way I want to set my system up, I want to prompt the user
to enter a 1 if they know the extension, or a 2 for a directory and
nothing else.
It works, however there is a 5 to 10 second delay after enter the 1 or 2
before the system responds.
I have read over the wiki on how asterisk handles digit inputs, but
cannot seem to isolate the problem. No other extension beginning with
(or even including) a '1' or a '2'.
Is this just how the system operates, or am I missing something..?
Thanks !!
Here is the [start] in my extensions.conf :
[start]
; If someone dials the Operator, just start them here.
exten = 0,1,Goto(s,1)
exten = s,1,Wait,1 ; Wait a second, just for fun
exten = s,2,Answer ; Answer the line
exten = s,3,SetMusicOnHold,default
exten = s,4,ResponseTimeout,5 ; Set Response Timeout
; Is is Morning, Afternoon or Evening ?
; Lets play a differnet greeting for each time period.
exten = s,5,AGI(openclose.agi)
exten = s,6,GotoIF($[${STATUS} = morning]?10)
exten = s,7,GotoIF($[${STATUS} = afternoon]?12)
exten = s,8,GotoIF($[${STATUS} = evening]?14)
extex = s,9,Goto(s,6)
; The various Greetings based on Time of Day
exten = s,10,Background(rjs-morning-welcome)
exten = s,11,Goto(s,15)
exten = s,12,Background(rjs-afternoon-welcome)
exten = s,13,Goto(s,15)
exten = s,14,Background(rjs-evening-welcome)
; The Voice Menu
exten = s,15,Background(rjs-if-you-know-the-extension)
exten = s,16,Wait,1
exten = s,17,BackGround(to-dial-by-name-press) ; Play some instructions
exten = s,18,BackGround(digits/2) ; Play some instructions
; A timeout and invalid extension rule
;
exten = t,1,Goto(s,15) ; If they take too long, give up
exten = i,1,Playback(invalid) ; That's not valid, try again
; If they know the extension, send them on.
exten = 1,1,Goto(extension_is_known,s,1)
; Allow users the ability to get Directory listing (user must be in
voicemail.conf)
exten = 2,1,Directory,default|internal_extensions
here is the output of my show dialplan start:
pbx01*CLI show dialplan start
[ Context 'start' created by 'pbx_config' ]
'0' =1. Goto(s|1) [pbx_config]
'1' =1. Goto(extension_is_known|s|1) [pbx_config]
'2' =1. Directory(default|internal_extensions) [pbx_config]
'i' =1. Playback(invalid) [pbx_config]
's' =1. Wait(1)[pbx_config]
2. Answer() [pbx_config]
3. SetMusicOnHold(default)[pbx_config]
4. ResponseTimeout(5) [pbx_config]
5. AGI(openclose.agi) [pbx_config]
6. GotoIF($[${STATUS} = morning]?10) [pbx_config]
7. GotoIF($[${STATUS} = afternoon]?12)[pbx_config]
8. GotoIF($[${STATUS} = evening]?14) [pbx_config]
10. Background(rjs-morning-welcome) [pbx_config]
11. Goto(s|15)[pbx_config]
12. Background(rjs-afternoon-welcome) [pbx_config]
13. Goto(s|15)[pbx_config]
14. Background(rjs-evening-welcome) [pbx_config]
15. Background(rjs-if-you-know-the-extension) [pbx_config]
16. Wait(1) [pbx_config]
17. BackGround(to-dial-by-name-press) [pbx_config]
18. BackGround(digits/2) [pbx_config]
't' =1. Goto(s|15) [pbx_config]
**
Richard J. Sears
Vice President
American Internet Services
[EMAIL PROTECTED]
http://www.adnc.com
858.576.4272 - Phone
858.427.2401 - Fax
INOC-DBA - 6130
I fly because it releases my mind
from the tyranny of petty things . .
Work like you don't need the money, love like you've
never been hurt and dance like you do when nobody's
watching.
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[Asterisk-Users] Which Codec(s) to use..?
Hey Everyone, I am playing around with my * box, and I have a few different phones hanging off it it right now. I have a Cisco 7960 capable of g729, ulaw and alaw, I have a Cisco ATA186 with a Panasonic cordless phone attached to it, I have a Digum IAXy with a dumb analog phone attached to it, and I have a Linksys PAP2-NA with an ATT 959 analog phone attached to it. I also have several IAX2 connections, one to NuFone and one to another provider. My question revolves around which codec to use. I purchased 10 licenses of the g729 from Digum thinking it was the best since it costs money :-) I have read the wiki on the codecs, but I did not find any real practical use suggestions, just info on the codecs and the bandwidth they used. In several cases I am not worried about the bandwidth consumption as I control the connectivity (several phone are behind T1's or high speed DSL), one of the IAX connections is across my OC48, and several phones are sitting on my desktop on the lan connected to the * server. So my question is this - if I am running g729, am I giving up quality for the sake of bandwidth conservation..? Should I use the codec that matches the greatest amount of bandwidth that I have available, or go with the codec that uses the least amount of bandwidth..? Thanks ** Richard J. Sears Vice President American Internet Services [EMAIL PROTECTED] http://www.adnc.com 858.576.4272 - Phone 858.427.2401 - Fax INOC-DBA - 6130 I fly because it releases my mind from the tyranny of petty things . . Work like you don't need the money, love like you've never been hurt and dance like you do when nobody's watching. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Delay after entering digits with IVR
Yes, I am trying to collect digits during the playback of a file. I have
tried both background and backgrounddetect with the same result.
Basically I have a menu - Press 1 if you know your parties extension,
press 2 for a directory.
In either case (background or backgrounddetect) when I hit 1 or 2 there
is a 5 to 8 second delay AFTER I hit the button before it goes to the
menu.
On Thu, 24 Feb 2005 19:39:35 -0500
Race Vanderdecken [EMAIL PROTECTED] wrote:
Hmmm,
Are you trying to collect digits during a playback that is not set to
listen for a digit?
From the coding side I know that depending on how the prompt is called
you can enter a digit and interrupt the prompt. Otherwise the prompt
will finish and then see the digits.
What you want is BackgroundDetect(filename[|sil[|min|max]])
http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+BackGroundDete
ct
This will see the digits, and stop if it is one you have set.
Race The Tyrant Vandedecken
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Richard J.
Sears
Sent: Thursday, February 24, 2005 6:49 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Delay after entering digits with IVR
I have a [start] context that all my inbound and '0' calls are routed
into.
Because of the way I want to set my system up, I want to prompt the user
to enter a 1 if they know the extension, or a 2 for a directory and
nothing else.
It works, however there is a 5 to 10 second delay after enter the 1 or 2
before the system responds.
I have read over the wiki on how asterisk handles digit inputs, but
cannot seem to isolate the problem. No other extension beginning with
(or even including) a '1' or a '2'.
Is this just how the system operates, or am I missing something..?
Thanks !!
Here is the [start] in my extensions.conf :
[start]
; If someone dials the Operator, just start them here.
exten = 0,1,Goto(s,1)
exten = s,1,Wait,1 ; Wait a second, just for fun
exten = s,2,Answer ; Answer the line
exten = s,3,SetMusicOnHold,default
exten = s,4,ResponseTimeout,5 ; Set Response Timeout
; Is is Morning, Afternoon or Evening ?
; Lets play a differnet greeting for each time period.
exten = s,5,AGI(openclose.agi)
exten = s,6,GotoIF($[${STATUS} = morning]?10)
exten = s,7,GotoIF($[${STATUS} = afternoon]?12)
exten = s,8,GotoIF($[${STATUS} = evening]?14)
extex = s,9,Goto(s,6)
; The various Greetings based on Time of Day
exten = s,10,Background(rjs-morning-welcome)
exten = s,11,Goto(s,15)
exten = s,12,Background(rjs-afternoon-welcome)
exten = s,13,Goto(s,15)
exten = s,14,Background(rjs-evening-welcome)
; The Voice Menu
exten = s,15,Background(rjs-if-you-know-the-extension)
exten = s,16,Wait,1
exten = s,17,BackGround(to-dial-by-name-press) ; Play some
instructions
exten = s,18,BackGround(digits/2) ; Play some instructions
; A timeout and invalid extension rule
;
exten = t,1,Goto(s,15) ; If they take too long, give
up
exten = i,1,Playback(invalid) ; That's not valid, try again
; If they know the extension, send them on.
exten = 1,1,Goto(extension_is_known,s,1)
; Allow users the ability to get Directory listing (user must be in
voicemail.conf)
exten = 2,1,Directory,default|internal_extensions
here is the output of my show dialplan start:
pbx01*CLI show dialplan start
[ Context 'start' created by 'pbx_config' ]
'0' =1. Goto(s|1)
[pbx_config]
'1' =1. Goto(extension_is_known|s|1)
[pbx_config]
'2' =1. Directory(default|internal_extensions)
[pbx_config]
'i' =1. Playback(invalid)
[pbx_config]
's' =1. Wait(1)
[pbx_config]
2. Answer()
[pbx_config]
3. SetMusicOnHold(default)
[pbx_config]
4. ResponseTimeout(5)
[pbx_config]
5. AGI(openclose.agi)
[pbx_config]
6. GotoIF($[${STATUS} = morning]?10)
[pbx_config]
7. GotoIF($[${STATUS} = afternoon]?12)
[pbx_config]
8. GotoIF($[${STATUS} = evening]?14)
[pbx_config]
10. Background(rjs-morning-welcome)
[pbx_config]
11. Goto(s|15)
[pbx_config]
12. Background(rjs-afternoon-welcome)
[pbx_config]
13. Goto(s|15)
[pbx_config]
14. Background(rjs-evening-welcome)
[pbx_config]
15. Background(rjs-if-you-know-the-extension)
[pbx_config]
16. Wait(1)
[pbx_config]
17. BackGround(to-dial-by-name-press)
[pbx_config]
18. BackGround(digits/2)
[pbx_config]
't' =1. Goto(s|15)
[pbx_config]
**
Richard J. Sears
Vice President
Re: [Asterisk-Users] Send outgoing calls to a SIP gateway
Kanishka -
Just add something like this for an outbound context - in this case, I
match any calls dialed with 11 digits and point it out my NuFone
connection.
[pstn_outbound]
exten = _1NXXNXX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN})
and include it where you need it...
[default]
include = pstn_outbound
Be careful where you put it as it determines who has access to your
outbound dialtone.
On Wed, 23 Feb 2005 16:28:11 -
Kanishka Somaratne [EMAIL PROTECTED] wrote:
How do I route all the outgoing calls through a SIP gateway, this should send
more than one outgoing call to the sip gateway at once. please show me the
sample configurations on how to do this.
my SIP gatway can accecpt direct IP traffic or SIP proxy traffc.
Thank You
Kanishka
**
Richard J. Sears
Vice President
American Internet Services
[EMAIL PROTECTED]
http://www.adnc.com
858.576.4272 - Phone
858.427.2401 - Fax
INOC-DBA - 6130
I fly because it releases my mind
from the tyranny of petty things . .
Work like you don't need the money, love like you've
never been hurt and dance like you do when nobody's
watching.
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Re: [Asterisk-Users] Able to tell if phone is registered?
One method that I use is to dial all of my extensions at once (Office,
home, laptop) and then dial my cellphone. Failing any of those to answer,
it drops back into my voicemail.
exten =
${RJSWORK},1,Macro(stdexten_cell,SIP/${RJSWORK}SIP/${RJSDESK}SIP/${RJSLAPTOP}SIP/${RJSHOME},${RJSCELL})
[macro-stdexten_cell]
exten = s,1,Dial(${ARG1},10,rtm)
exten = s,2,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Dial(IAX2/[EMAIL PROTECTED]/${ARG2},10,rtm)
exten = s-NOANSWER,2,Voicemail(u${MACRO_EXTEN})
;exten = s-NOANSWER,3,Goto(default,s,1)
exten = s-BUSY,1,Voicemail(b${MACRO_EXTEN})
;exten = s-BUSY,2,Goto(default,s,1)
exten = s-.,1,Goto(s-NOANSWER,1)
exten = a,1,VoicemailMain(${MACRO_EXTEN})
On Wed, 23 Feb 2005 12:07:03 -0600
Nathan C. Smith [EMAIL PROTECTED] wrote:
Hi All,
I have a new asterisk setup running at home and am very happy with it.
One thing that I am trying to do is to take various actions in the
dialplan *if* a particular phone is registered/authenticated/connected.
For example, if someone dials *me* and is shows that I am connected via
my softphone, to try it instead of my deskphone (and possibly notifiy
the user in advance that it is taking that action).
There are a couple ways, but within the dialplan you can use chanavail().
On the console you can type show application appname to get more
information about applications such as chanavail and dial.
On the failure of a dial command the priority jumps 101, so for instance if
you softphone was dialed at priority 3 and it failed the execution would
jump to priority 104 where you could attempt to dial your deskphone.
Finally, you can use the asterisk database. By dialing an extension to set
a database value and checking for a value in the dialplan you can use logic
to decide whether to dial an extension or not.
Most of this is documented reasonably well on www.voip-info.org once you
know what you are looking for.
-Nate
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**
Richard J. Sears
Vice President
American Internet Services
[EMAIL PROTECTED]
http://www.adnc.com
858.576.4272 - Phone
858.427.2401 - Fax
INOC-DBA - 6130
I fly because it releases my mind
from the tyranny of petty things . .
Work like you don't need the money, love like you've
never been hurt and dance like you do when nobody's
watching.
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[Asterisk-Users] Extension Design in Visio
Hey Everyone - I was going to create a visio diagram outlining how my extensions will flow out. I was just wondering if anyone on the list may have an example they have already done up so I can get some ideas. Thanks ** Richard J. Sears Vice President American Internet Services [EMAIL PROTECTED] http://www.adnc.com 858.576.4272 - Phone 858.427.2401 - Fax INOC-DBA - 6130 I fly because it releases my mind from the tyranny of petty things . . Work like you don't need the money, love like you've never been hurt and dance like you do when nobody's watching. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Cisco 7970 Won't boot after factory reset
Hi Keith, I have a TFTP server set up with the proper files on it, but after a factory reset, how does the phone know where to find the TFTP server..? I cannot get into it to set the TFTP server IP address. Thanks On Wed, 16 Feb 2005 20:02:22 -0500 Keith O'Brien [EMAIL PROTECTED] wrote: It is trying to download its firmware. You need to setup a TFTP Server. Also be aware that the 7970 only supports SCCP not SIP. Further, the * implementation of SCCP doesn't support the latest version of SCCP which is required for the 7970. I don't see how it would work at all with *. Hi Everyone - I just got my hands on a Cisco 7970 and was told that I should do a factory reset before trying to configure it to work with Asterisk. After the factory reset, it will not boot at all, instead sits with the line button lights flashing one at a time in sequence. I have had no luck trying to figure it out - anyone run into the same problem that can lend a hand..? ** Richard J. Sears Vice President American Internet Services [EMAIL PROTECTED] http://www.adnc.com 858.576.4272 - Phone 858.427.2401 - Fax INOC-DBA - 6130 I fly because it releases my mind from the tyranny of petty things . . Work like you don't need the money, love like you've never been hurt and dance like you do when nobody's watching. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Cisco 7970 Won't boot after factory rese t
Duh !! That did it :-) Thanks On Thu, 17 Feb 2005 10:22:03 -0700 Colin Anderson [EMAIL PROTECTED] wrote: how does the phone know where to find the TFTP server..? Dude, option 150 in your DHCP server: http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_tech_note09186 a00800942f4.shtml We use the same option for our Mitel phones. HTH. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ** Richard J. Sears Vice President American Internet Services [EMAIL PROTECTED] http://www.adnc.com 858.576.4272 - Phone 858.427.2401 - Fax INOC-DBA - 6130 I fly because it releases my mind from the tyranny of petty things . . Work like you don't need the money, love like you've never been hurt and dance like you do when nobody's watching. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAXy Provisioning Using Windows
I just used this on two brand new IAX devices !! It worked like a charm !! Great Job ! On Thu, 17 Feb 2005 10:37:16 -0700 Tony da Costa [EMAIL PROTECTED] wrote: For anyone playing around with IAXy(S100i) devices, I am making the following available: Windows IAXy Provision v1.00 This is a from-the-ground-up development of a means of provisioning IAXy devices using a Windows environment. For some users, being bound to Linux for IAXy provisioning is not viable or convenient in some cases. This application provides a GUI data entry for the various IAXy parameters and communicates the new parameters to the selected IAXy. You are free to do with this application as you wish. It is provided as-is with the hope that it will make someone's day a little easier. A download package is available at: http://dacosta.dynip.com/asterisk ...Tony da Costa ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ** Richard J. Sears Vice President American Internet Services [EMAIL PROTECTED] http://www.adnc.com 858.576.4272 - Phone 858.427.2401 - Fax INOC-DBA - 6130 I fly because it releases my mind from the tyranny of petty things . . Work like you don't need the money, love like you've never been hurt and dance like you do when nobody's watching. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco 7970 Won't boot after factory reset
Hi Everyone - I just got my hands on a Cisco 7970 and was told that I should do a factory reset before trying to configure it to work with Asterisk. After the factory reset, it will not boot at all, instead sits with the line button lights flashing one at a time in sequence. I have had no luck trying to figure it out - anyone run into the same problem that can lend a hand..? Thanks ** Richard J. Sears Vice President American Internet Services [EMAIL PROTECTED] http://www.adnc.com 858.576.4272 - Phone 858.427.2401 - Fax INOC-DBA - 6130 I fly because it releases my mind from the tyranny of petty things . . Work like you don't need the money, love like you've never been hurt and dance like you do when nobody's watching. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TFTP Serer ????
Is the Cisco phone book via XML something specific to [EMAIL PROTECTED], or is this something that can be implemented within a normal Asterisk deployment..? Thanks On Mon, 14 Feb 2005 17:43:36 -0800 (PST) [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: [EMAIL PROTECTED] has Asterisk, a TFTP server, and a web based cisco phone config tool. It auto installs it all. This should save you a lot of time. you can be up and running in an hour. It also has a built in phone book cisco XML service that works well with the 7960. http://asteriskathome.sourceforge.net/ --- Stefan Gofferje [EMAIL PROTECTED] wrote: Ferguson, Michael schrieb: G'Day All, Can someone help me out please. My new CISCO 7960's manual says I have to setup a TFTP server. Googled it and got a little understanding, but from * standpoint, well I am still a lost. Can I set this tftp server on the same * box? Can in be on a WinXP box? Which tftp software would you recommend? Any Linux distro should ship with one or two tftp servers. Anyway, away from firmware updates, the config could be done via phone menu or webinterface. There also are various tftpds available for Windows. BTY: Does anyone have a How-To on getting the 7960 fully configured for *? http://www.voip-info.org/tiki-index.php?page=cisco%2079xx http://www.voip-info.org/tiki-index.php?page=Setup%20SiP%20on%207940%20-%207960 Regards, Stefan -- (o_ Stefan Gofferje | Linux Systems Specialist //\ Reg'd Linux User #247167 | Network Security Specialist V_/_ Linux is like a Wigwam - No gates, no windows, Apache inside ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ** Richard J. Sears Vice President American Internet Services [EMAIL PROTECTED] http://www.adnc.com 858.576.4272 - Phone 858.427.2401 - Fax INOC-DBA - 6130 I fly because it releases my mind from the tyranny of petty things . . Work like you don't need the money, love like you've never been hurt and dance like you do when nobody's watching. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] X-Lite Softphone
Hey Everyone, I downloaded and installed the X-Lite softphone the other day (the lite version) and cannot seem to get it to work well. Don't get me wrong, it registers with my asterisk server and everything seems to work well, except the call quality really is horrible. I thought it may be the place I was trying it at (DSL) so I took it to the office and tried it right next to the asterisk box and had the same luck. My laptop is the Dell XPS, so power, ram, etc are not problems, and loading it onto my desktop system revealed the same results. There was also no difference between a NAT implementation and a regular (live IP) implementation of the software. I am getting stuttering speech, cutouts, etc all the time. Running my Cisco 7960 at the same locations and it works fantastic with no issues at all. Is anyone else using this softphone or does anyone know of a better softphone or some hints on configuration that may make X-Lite work better..? TIA ** Richard J. Sears Vice President American Internet Services [EMAIL PROTECTED] http://www.adnc.com 858.576.4272 - Phone 858.427.2401 - Fax INOC-DBA - 6130 I fly because it releases my mind from the tyranny of petty things . . Work like you don't need the money, love like you've never been hurt and dance like you do when nobody's watching. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X-Lite Softphone
Thanks Liaan, Yes - that was the first thing I checked. My laptop is running an XPS 3.4Ghz Extreme Processor with 1GB RAM, neither CPU nor memory even flinched while the calls were in session. I switched to a desktop thinking just maybe it didn't like my laptop with XP Pro on it. :-) On Tue, 15 Feb 2005 09:22:53 -0800 (PST) Liaan vd Merwe [EMAIL PROTECTED] wrote: Hi while on a call.. did you check your CPU usage.. i have a P3 and sometimes when i move my mouse, xlite starts to stutter.. cpu then running 100% just my 2cents chow L - Original Message - From: Richard J. Sears [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Tuesday, February 15, 2005 6:56 PM Subject: [Asterisk-Users] X-Lite Softphone Hey Everyone, I downloaded and installed the X-Lite softphone the other day (the lite version) and cannot seem to get it to work well. Don't get me wrong, it registers with my asterisk server and everything seems to work well, except the call quality really is horrible. I thought it may be the place I was trying it at (DSL) so I took it to the office and tried it right next to the asterisk box and had the same luck. My laptop is the Dell XPS, so power, ram, etc are not problems, and loading it onto my desktop system revealed the same results. There was also no difference between a NAT implementation and a regular (live IP) implementation of the software. I am getting stuttering speech, cutouts, etc all the time. Running my Cisco 7960 at the same locations and it works fantastic with no issues at all. Is anyone else using this softphone or does anyone know of a better softphone or some hints on configuration that may make X-Lite work better..? TIA ** Richard J. Sears Vice President American Internet Services [EMAIL PROTECTED] http://www.adnc.com 858.576.4272 - Phone 858.427.2401 - Fax INOC-DBA - 6130 I fly because it releases my mind from the tyranny of petty things . . Work like you don't need the money, love like you've never been hurt and dance like you do when nobody's watching. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ** Richard J. Sears Vice President American Internet Services [EMAIL PROTECTED] http://www.adnc.com 858.576.4272 - Phone 858.427.2401 - Fax INOC-DBA - 6130 I fly because it releases my mind from the tyranny of petty things . . Work like you don't need the money, love like you've never been hurt and dance like you do when nobody's watching. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X-Lite Softphone
I tried both with and without a headset. On Tue, 15 Feb 2005 09:32:39 -0800 Robert Goodyear [EMAIL PROTECTED] wrote: Hey Everyone, I downloaded and installed the X-Lite softphone the other day (the lite version) and cannot seem to get it to work well. Are you using a headset? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ** Richard J. Sears Vice President American Internet Services [EMAIL PROTECTED] http://www.adnc.com 858.576.4272 - Phone 858.427.2401 - Fax INOC-DBA - 6130 I fly because it releases my mind from the tyranny of petty things . . Work like you don't need the money, love like you've never been hurt and dance like you do when nobody's watching. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X-Lite Softphone
Hi Wiley - Are you running the free version..? Is there a quality diff in the free vs. pay versions..? Thanks for all the info. On Tue, 15 Feb 2005 11:14:25 -0700 Wiley Siler [EMAIL PROTECTED] wrote: I recently just started using this phone again and it works very well. Call quality is excellent for me. I even have a user that VPNs into my network and attaches to Asterisk box with X-lite. Call quality is still excellent despite extra overhead from VPN. In case you did not have this link... http://www.voip-info.org/tiki-index.php?page=Asterisk%20phone%20xten%20x lite Your codec selection will also affect your perceived call quality. Are you using GSM or ULAw? I use uLaw on all internal connected softphones. Why not? We have the internal bandwidth. In short, there is certainly nothing wrong with the software. I would check... Codec selections - X-lite (deselect all but G711u) and in Asterisk configs (bandwidth = high is set?) X-Lite configuration exactly as in the link above... Headset is of good quality? (I use a $70 Plantronics USB headset and it works great) Are you running a ton of junk in your system tray? Do you have a billion little memory eaters loaded that you don't even use? You would be shocked at how many people don't realize how many junk items run in the system tray and eat memory. Drop things like, winamp agent, quicktime agent, winzip quickstart, etc, etc, etc I use my softphone as my primary phone now. All calls to my Polycom IP500 forward to my PC. I would never go back either. It is just too convenient to pause the MP3 player and take the call handsfree. Luck, Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Richard J. Sears Sent: Tuesday, February 15, 2005 10:42 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] X-Lite Softphone I tried both with and without a headset. On Tue, 15 Feb 2005 09:32:39 -0800 Robert Goodyear [EMAIL PROTECTED] wrote: Hey Everyone, I downloaded and installed the X-Lite softphone the other day (the lite version) and cannot seem to get it to work well. Are you using a headset? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ** Richard J. Sears Vice President American Internet Services [EMAIL PROTECTED] http://www.adnc.com 858.576.4272 - Phone 858.427.2401 - Fax INOC-DBA - 6130 I fly because it releases my mind from the tyranny of petty things . . Work like you don't need the money, love like you've never been hurt and dance like you do when nobody's watching. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ** Richard J. Sears Vice President American Internet Services [EMAIL PROTECTED] http://www.adnc.com 858.576.4272 - Phone 858.427.2401 - Fax INOC-DBA - 6130 I fly because it releases my mind from the tyranny of petty things . . Work like you don't need the money, love like you've never been hurt and dance like you do when nobody's watching. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ztdummy on Gentoo 2.6.10 Box
Hi Everyone, I read through the list on the issues with the ztdummy driver which I need for MeetMe, but I seem to have come across a problem that I cannot seem to find an answer for. I am running Gentoo 2.6.10 on an Intel box. I have read the the wiki entries on the ztdummy and followed the instructions as they relate to the 2.6 kernel. Everything compiled great, but a modprobe ztdummy returned this: WARNING: Error inserting zaptel (/lib/modules/2.6.10-gentoo-r6/misc/zaptel.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting zaptel (/lib/modules/2.6.10-gentoo-r6/misc/zaptel.ko): Unknown symbol in module, or unknown parameter (see dmesg) FATAL: Error inserting ztdummy (/lib/modules/2.6.10-gentoo-r6/misc/ztdummy.ko): Unknown symbol in module, or unknown parameter (see dmesg) FATAL: Error running install command for ztdummy Here is what my dmesg says: zaptel: Unknown symbol crc_ccitt_table zaptel: Unknown symbol crc_ccitt_table ztdummy: Unknown symbol zt_receive ztdummy: Unknown symbol zt_transmit ztdummy: Unknown symbol zt_unregister ztdummy: Unknown symbol zt_register I DO NOT have any Zaptel devices in my system and I VIed the makefile and uncommented the ztdummy as instructed. I guess my question is - what the heck is happeneing. Why is ztdummy trying to load zaptel which I do not have in my system..? Any help would be greatly appreciated. Thanks ** Richard J. Sears Vice President American Internet Services [EMAIL PROTECTED] http://www.adnc.com 858.576.4272 - Phone 858.427.2401 - Fax INOC-DBA - 6130 I fly because it releases my mind from the tyranny of petty things . . Work like you don't need the money, love like you've never been hurt and dance like you do when nobody's watching. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ztdummy on Gentoo 2.6.10 Box
Hi Guills, I followed the instructions exactly as described on the voip pages as well as your instructions below. I still get the same errors. pbx01 zaptel # modprobe zaptel FATAL: Error inserting zaptel (/lib/modules/2.6.10-gentoo-r6/misc/zaptel.ko): Unknown symbol in module, or unknown parameter (see dmesg) DMESG: zaptel: Unknown symbol crc_ccitt_table And of course ztdummy fails to load becuase the zaptel does not load. Thanks On Mon, 14 Feb 2005 13:55:04 -0800 Chamberland-Larose, Guillaume [EMAIL PROTECTED] wrote: The ztdummy requires zaptel to be loaded. You don't need any zaptel devices just the kernel module. Get zaptel then follow the instructions on the wiki @ http://www.voip-info.org/wiki-Asterisk+timer+ztdummy i.e.: ztdummy and Kernel 2.6 Recent CVS versions (as of July 2004) also have zaptel modules which work with 2.6 kernels (type make linux26 to compile them). The 2.6 version of the ztdummy module is completely different to the 2.4 version. Firstly it does not rely on the USB hardware being there; instead it uses the PC hardware's clock (which under 2.6 kernels can be set to generate interrupts at the required precision), so it can be used on any machine, rather than only those with the right USB hardware. Effectively the 2.6 version of ztdummy does the same job as zaprtc does for 2.4 kernels. I have been using ztdummy for 2.6 for a while now, and have had no problems with it. To install, simply checkout zaptel from the Asterisk CVS and do the following: - cd /usr/src/zaptel * READ /usr/src/zaptel/README.udev and follow the steps * check modules on: /etc/sysconfig/zaptel if you have no digium hardware comment out all modeules except ztdummy. - make linux26 - modprobe zaptel - modprobe ztdummy Cheers, Guills -Original Message- From: Richard J. Sears [mailto:[EMAIL PROTECTED] Sent: Monday, February 14, 2005 1:16 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] ztdummy on Gentoo 2.6.10 Box Hi Everyone, I read through the list on the issues with the ztdummy driver which I need for MeetMe, but I seem to have come across a problem that I cannot seem to find an answer for. I am running Gentoo 2.6.10 on an Intel box. I have read the the wiki entries on the ztdummy and followed the instructions as they relate to the 2.6 kernel. Everything compiled great, but a modprobe ztdummy returned this: WARNING: Error inserting zaptel (/lib/modules/2.6.10-gentoo-r6/misc/zaptel.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting zaptel (/lib/modules/2.6.10-gentoo-r6/misc/zaptel.ko): Unknown symbol in module, or unknown parameter (see dmesg) FATAL: Error inserting ztdummy (/lib/modules/2.6.10-gentoo-r6/misc/ztdummy.ko): Unknown symbol in module, or unknown parameter (see dmesg) FATAL: Error running install command for ztdummy Here is what my dmesg says: zaptel: Unknown symbol crc_ccitt_table zaptel: Unknown symbol crc_ccitt_table ztdummy: Unknown symbol zt_receive ztdummy: Unknown symbol zt_transmit ztdummy: Unknown symbol zt_unregister ztdummy: Unknown symbol zt_register I DO NOT have any Zaptel devices in my system and I VIed the makefile and uncommented the ztdummy as instructed. I guess my question is - what the heck is happeneing. Why is ztdummy trying to load zaptel which I do not have in my system..? Any help would be greatly appreciated. Thanks ** Richard J. Sears Vice President American Internet Services [EMAIL PROTECTED] http://www.adnc.com 858.576.4272 - Phone 858.427.2401 - Fax INOC-DBA - 6130 I fly because it releases my mind from the tyranny of petty things . . Work like you don't need the money, love like you've never been hurt and dance like you do when nobody's watching. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ** Richard J. Sears Vice President American Internet Services [EMAIL PROTECTED] http://www.adnc.com
Re: [Asterisk-Users] ztdummy on Gentoo 2.6.10 Box
Hi Guills, The module was actually already done, just not loaded !! Many thanks for the help on this one!! On Mon, 14 Feb 2005 16:29:31 -0800 Chamberland-Larose, Guillaume [EMAIL PROTECTED] wrote: This last error message now means zaptel is compiled (which it wasn't before) but now it can't find that table. You need to enable it in your kernel config it seems. As mentionned here: http://lists.digium.com/pipermail/asterisk-dev/2004-December/008303.html The option to turn this on is in the last submenu of the kernel config menu. Guills -Original Message- From: Richard J. Sears [mailto:[EMAIL PROTECTED] Sent: Monday, February 14, 2005 4:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Chamberland-Larose, Guillaume Subject: Re: [Asterisk-Users] ztdummy on Gentoo 2.6.10 Box Hi Guills, I followed the instructions exactly as described on the voip pages as well as your instructions below. I still get the same errors. pbx01 zaptel # modprobe zaptel FATAL: Error inserting zaptel (/lib/modules/2.6.10-gentoo-r6/misc/zaptel.ko): Unknown symbol in module, or unknown parameter (see dmesg) DMESG: zaptel: Unknown symbol crc_ccitt_table And of course ztdummy fails to load becuase the zaptel does not load. Thanks On Mon, 14 Feb 2005 13:55:04 -0800 Chamberland-Larose, Guillaume [EMAIL PROTECTED] wrote: The ztdummy requires zaptel to be loaded. You don't need any zaptel devices just the kernel module. Get zaptel then follow the instructions on the wiki @ http://www.voip-info.org/wiki-Asterisk+timer+ztdummy i.e.: ztdummy and Kernel 2.6 Recent CVS versions (as of July 2004) also have zaptel modules which work with 2.6 kernels (type make linux26 to compile them). The 2.6 version of the ztdummy module is completely different to the 2.4 version. Firstly it does not rely on the USB hardware being there; instead it uses the PC hardware's clock (which under 2.6 kernels can be set to generate interrupts at the required precision), so it can be used on any machine, rather than only those with the right USB hardware. Effectively the 2.6 version of ztdummy does the same job as zaprtc does for 2.4 kernels. I have been using ztdummy for 2.6 for a while now, and have had no problems with it. To install, simply checkout zaptel from the Asterisk CVS and do the following: - cd /usr/src/zaptel * READ /usr/src/zaptel/README.udev and follow the steps * check modules on: /etc/sysconfig/zaptel if you have no digium hardware comment out all modeules except ztdummy. - make linux26 - modprobe zaptel - modprobe ztdummy Cheers, Guills -Original Message- From: Richard J. Sears [mailto:[EMAIL PROTECTED] Sent: Monday, February 14, 2005 1:16 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] ztdummy on Gentoo 2.6.10 Box Hi Everyone, I read through the list on the issues with the ztdummy driver which I need for MeetMe, but I seem to have come across a problem that I cannot seem to find an answer for. I am running Gentoo 2.6.10 on an Intel box. I have read the the wiki entries on the ztdummy and followed the instructions as they relate to the 2.6 kernel. Everything compiled great, but a modprobe ztdummy returned this: WARNING: Error inserting zaptel (/lib/modules/2.6.10-gentoo-r6/misc/zaptel.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting zaptel (/lib/modules/2.6.10-gentoo-r6/misc/zaptel.ko): Unknown symbol in module, or unknown parameter (see dmesg) FATAL: Error inserting ztdummy (/lib/modules/2.6.10-gentoo-r6/misc/ztdummy.ko): Unknown symbol in module, or unknown parameter (see dmesg) FATAL: Error running install command for ztdummy Here is what my dmesg says: zaptel: Unknown symbol crc_ccitt_table zaptel: Unknown symbol crc_ccitt_table ztdummy: Unknown symbol zt_receive ztdummy: Unknown symbol zt_transmit ztdummy: Unknown symbol zt_unregister ztdummy: Unknown symbol zt_register I DO NOT have any Zaptel devices in my system and I VIed the makefile and uncommented the ztdummy as instructed. I guess my question is - what the heck is happeneing. Why is ztdummy trying to load zaptel which I do not have in my system..? Any help would be greatly appreciated. Thanks ** Richard J. Sears Vice President American Internet Services [EMAIL PROTECTED] http://www.adnc.com
