Re: [asterisk-users] stopping unwanted attempts
I see MANY of these in my log files: [Jan 15 03:06:12] NOTICE[14129] chan_sip.c: Registration from '202 sip:202@X:5060' failed for '37.8.12.147:26832' - Wrong password [Jan 15 03:06:19] NOTICE[14129] chan_sip.c: Registration from '5001 sip:5001@X:5060' failed for '37.8.12.147:21268' - Wrong password [Jan 15 03:06:23] NOTICE[14129] chan_sip.c: Registration from '30 sip:30@X:5060' failed for '37.8.12.147:21270' - Wrong password [Jan 15 03:06:48] NOTICE[14129] chan_sip.c: Registration from '70 sip:70@X:5060' failed for '37.8.12.147:21328' - Wrong password [Jan 15 03:06:50] NOTICE[14129][C-0085] chan_sip.c: Call from '' ( 8.33.7.110:5103) to extension '889011972592735467' rejected because extension not found in context 'default'. [Jan 15 03:06:56] NOTICE[14129] chan_sip.c: Registration from '4 sip:4@X:5060' failed for '37.8.12.147:21272' - Wrong password [Jan 15 03:07:11] NOTICE[14129] chan_sip.c: Registration from '12001 sip:12001@X:5060' failed for '37.8.12.147:5060' - Wrong password [Jan 15 03:34:02] NOTICE[14129][C-0086] chan_sip.c: Call from '' ( 172.246.236.90:5078) to extension '8889011972595301123' rejected because extension not found in context 'default'. What is the correct way to block these idiots so they don't even get this far. Thanks, Jerry At this past year's AstriCon there was a series of security talks that covered fail2ban and best practices. You can view the playlist of videos on YouTube. The content should be helpful for you: https://www.youtube.com/playlist?list=PLighc-2vlRgT3DhE9DkIgSmpUX6v2AtYo Links to the playlists are also on asterisk.org: http://www.asterisk.org/community/astricon-user-conference/video-archive Cheers, Billy Chia -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to install asterisk in ubuntu?
There is a step-by-step tutorial with video on the Digium blog: http://blogs.digium.com/2012/11/14/how-to-install-asterisk-11-on-ubuntu-12-4-lts/ Additionally Emiliano's advice is excellent You can read Asterisk - The future of telephony and get a lot of stuff. Emiliano. However, I would recommend you read Asterisk the Definitive Guide. The Future of Telephony is now an outdated version of the book and the name has been changed to the Definitive Guide. In the modern version of the book there is installation instruction for both redhat-based and debian-based linux. *Billy Chia* -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How do I remotely force an *unconfigured* Digium DPMA
Apparently notify.check-sync does work but only if you're NOT using the DPMA. I just tried it and the phone just responds with a 200/OK and does nothing. Did you disable enable_check_sync in the xml config? By default this option is enabled and phones should restart with check-sync SIP NOTIFY https://wiki.asterisk.org/wiki/display/DIGIUM/Provisioning#Provisioning-RemoteRestart *Billy Chia* Digium, Inc. | Product Marketing Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA direct: +1 256-428-6099 *Check us out at*: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Custom Application recording problem
Hi Dale,
Thanks for the correction gosub() worked. There was a problem with pressing
option 3 so I removed extension 4.
Below is the final [sub-timo]
[sub-timo]
exten = s,1,Set(RecordingType=${ARG1})
exten = s,n,Set(TIMEOUT(digit)=2) ; Set Digit Timeout to 5
seconds
exten = s,n,Set(TIMEOUT(response)=2) ; Set Response Timeout to 10
seconds
exten = s,n,Answer
exten = s,n,NoOp(${CALLERID(num)})
exten = s,n,Set(number=${CALLERID(num)})
exten = s,n,NoOp(${number})
exten = s,n(recordmsg),Background(recmsg1) ;Please say yo message after
the beep and end with a hash
exten =
s,n,Record(/var/www/html/timo/crystalrecords/${RecordingType}/${number}.wav)
exten =
s,n(playmsg),Playback(/var/www/html/timo/crystalrecords/${RecordingType}/${n
umber})
exten = s,n(askuser),Background(ackrec) ;Press 1 to replay or 2 to
re-record, 3 to save
exten = s,11,WaitExten(5)
exten = 1,1,Goto(s,playmsg)
exten = 2,1,Goto(s,recordmsg) ; re-record message
exten = 3,1,AGI(${RecordingType}.php)
exten = s,1,Background(invalidentry)
exten = s,n,Goto(s,askuser)
exten = t,1,Playback(thankyoubye)
exten = t,n,Return
Inorder for the system to recognize invalid selections, I also changed
exten = i,1,Background(invalidentry)
exten = i,n,Goto(s,askuser)
To
exten = s,1,Background(invalidentry)
exten = s,n,Goto(s,askuser)
Thank you very much for the help.
Kind Regards
Billy
On 4/17/12 11:11 PM, Dale Noll dn...@wi.rr.com wrote:
Billy,
I really should have had my coffee before answering you previous
message. My head was in the wrong place (not saying where) and I sent
you down the wrong path.
Macro() is not the answer because of the WaitExten(). When WaitExten is
used in a Macro(), it does not match within the macro, it matches an
extension within the context where the macro was called. This is what
is causing your errors.
What you really should do is use gosub(), not macro().
Here is the recording routine
[sub-timo]
exten = s,1,Set(RecordingType=${ARG1})
exten = s,n,Set(TIMEOUT(digit)=2) ; Set Digit Timeout to 5
seconds
exten = s,n,Set(TIMEOUT(response)=2) ; Set Response Timeout to
10 seconds
exten = s,n,Answer
exten = s,n,NoOp(${CALLERID(num)})
exten = s,n,Set(number=${CALLERID(num)})
exten = s,n,NoOp(${number})
exten = s,n(recordmsg),Background(recmsg1) ;Please say yo message
after the beep and end with a hash
exten =
s,n,Record(/var/www/html/timo/crystalrecords/${RecordingType}/${number}.gsm)
exten =
s,n(playmsg),Playback(/var/www/html/timo/crystalrecords/${RecordingType}/${num
ber})
exten = s,n(askuser),Background(ackrec) ;Press 1 to replay or 2 to
re-record, 3 to save
exten = s,11,WaitExten(5)
exten = 1,1,Goto(s,playmsg)
exten = 2,1,Goto(s,recordmsg) ; re-record message
exten = 3,1,Goto(4,1)
exten = 4,1,AGI($RecordingType}.php)
exten = 4,n,Return()
exten = i,1,Background(invalidentry)
exten = i,n,Goto(s,askuser)
exten = t,1,Playback(thankyoubye)
exten = t,n,Return
I know big change there eh? Note: I did make some changes to extension
4, but that was fix syntax error, not because of the change from macro
to gosub.
The difference is really how you call it.
exten = 3552,1,Gosub(sub-timo,s,1(contentdb))
exten = 3552,n,Hangup()
Also note. I have not tested this code. I have something similar in
place, but not your specific code.
Oh. You should be able to remove the 'include = timo' from the
[from-internal-custom] context.
Dale
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Re: [asterisk-users] Custom Application recording problem
, ) in new
stack
-- Executing [s@macro-timo:5] NoOp(SIP/261-005c, 261) in new
stack
-- Executing [s@macro-timo:6] Set(SIP/261-005c, number=261) in
new stack
-- Executing [s@macro-timo:7] NoOp(SIP/261-005c, 261) in new
stack
-- Executing [s@macro-timo:8] BackGround(SIP/261-005c, recmsg1)
in new stack
-- SIP/261-005c Playing 'recmsg1.gsm' (language 'en')
-- Executing [s@macro-timo:9] Record(SIP/261-005c,
/var/www/html/timo/crystalrecords/contentdb/261.gsm) in new stack
-- SIP/261-005c Playing 'beep.gsm' (language 'en')
-- Executing [s@macro-timo:10] Playback(SIP/261-005c,
/var/www/html/timo/crystalrecords/contentdb/261) in new stack
-- SIP/261-005c Playing
'/var/www/html/timo/crystalrecords/contentdb/261.gsm' (language 'en')
-- Executing [s@macro-timo:11] BackGround(SIP/261-005c, ackrec)
in new stack
-- SIP/261-005c Playing 'ackrec.gsm' (language 'en')
-- Invalid extension '1' in context 'from-internal' on SIP/261-005c
== CDR updated on SIP/261-005c
-- Executing [i@from-internal:1] BackGround(SIP/261-005c,
invalidentry) in new stack
-- SIP/261-005c Playing 'invalidentry.slin' (language 'en')
-- Executing [i@from-internal:2] Goto(SIP/261-005c, 3589,2) in
new stack
-- Goto (from-internal,3589,2)
-- Executing [3589@from-internal:2] Read(SIP/261-005c,
choice,,1) in new stack
-- Accepting a maximum of 1 digits.
-- User entered nothing.
-- Executing [3589@from-internal:3] AGI(SIP/261-005c, rsvp.php|)
in new stack
-- Executing [3589@from-internal:4] Wait(SIP/261-005c, 1) in new
stack
-- Executing [3589@from-internal:5] Playback(SIP/261-005c,
silence/1cannot-complete-as-dialedcheck-number-dial-again,noanswer) in
new stack
-- SIP/261-005c Playing 'silence/1.gsm' (language 'en')
-- SIP/261-005c Playing 'cannot-complete-as-dialed.gsm' (language
'en')
-- SIP/261-005c Playing 'check-number-dial-again.gsm' (language
'en')
-- Executing [3589@from-internal:6] Wait(SIP/261-005c, 1) in new
stack
-- Executing [3589@from-internal:7] Congestion(SIP/261-005c, 20)
in new stack
== Spawn extension (from-internal, 3589, 7) exited non-zero on
'SIP/261-005c'
-- Executing [h@from-internal:1] Macro(SIP/261-005c, hangupcall)
in new stack
-- Executing [s@macro-hangupcall:1] GotoIf(SIP/261-005c,
1?noautomon) in new stack
-- Goto (macro-hangupcall,s,3)
-- Executing [s@macro-hangupcall:3] NoOp(SIP/261-005c,
TOUCH_MONITOR_OUTPUT=) in new stack
-- Executing [s@macro-hangupcall:4] GotoIf(SIP/261-005c,
1?noautomon2) in new stack
-- Goto (macro-hangupcall,s,6)
-- Executing [s@macro-hangupcall:6] NoOp(SIP/261-005c,
MONITOR_FILENAME=) in new stack
-- Executing [s@macro-hangupcall:7] GotoIf(SIP/261-005c,
1?skiprg) in new stack
-- Goto (macro-hangupcall,s,10)
-- Executing [s@macro-hangupcall:10] GotoIf(SIP/261-005c,
1?skipblkvm) in new stack
-- Goto (macro-hangupcall,s,13)
-- Executing [s@macro-hangupcall:13] GotoIf(SIP/261-005c,
1?theend) in new stack
-- Goto (macro-hangupcall,s,15)
-- Executing [s@macro-hangupcall:15] Hangup(SIP/261-005c, ) in
new stack
== Spawn extension (macro-hangupcall, s, 15) exited non-zero on
'SIP/261-005c' in macro 'hangupcall'
== Spawn extension (from-internal, h, 1) exited non-zero on
'SIP/261-005c'
-
Kind Regards
Billy
On 4/17/12 1:56 PM, Dale Noll dn...@wi.rr.com wrote:
On 04/16/2012 04:09 PM, Billy Kaye wrote:
Re: [asterisk-users] Custom Application recording problem Thanks Dale,
Am not sure why it was working in 1.4 but for some reason it was ( Note : My
Asterisk is running bundled with Elastix).
But any your suggestion worked very fine.
Glad to hear it.
Now am having one problem how can define those extensions only with in
different contexts, the problem I see is since am
Building 3 recording applications only one will be able call its AGI file,
Say if someone calls custom extension 1114
They can record message
-Press 1 to Replay Press 2 to Re-record or Press 3 to Save the file
Also if someone calls custom extension 1115
-Press 1 to Replay Press 2 to Re-record or Press 3 to Save the file
Note Each save file selection calls a different AGI file
E.g
exten = 1,1,Goto,timo|3552|9
exten = 2,1,Goto(3552,7) ; re-record message
exten = 3,1,Goto(4,1)
exten = 4,AGI(timorec.php)
There a few ways to do it. Probably the easiest to maintain in the long run
would be via the use of a macro.
[macro-timo]
exten = s,1,Set(RecordingType=${ARG1})
exten = s,n,Set(TIMEOUT(digit)=2) ; Set Digit Timeout to 5
seconds
exten = s,n,Set(TIMEOUT(response)=2) ; Set Response Timeout to 10
seconds
exten = s,n,Answer
exten = s,n,NoOp(${CALLERID(num)})
exten = s,n,Set(number=${CALLERID(num)})
exten = s,n,NoOp
[asterisk-users] Custom Application recording problem
Greetings All,
I have a compatibilty problem between asterisk 1.4 and 1.6.2
In my 1.4 asterisk I have a custom application that users call and make
recordings which recording I save to a file with the caller Id.
Below is the config file which works perfectly in 1.4
[timo]
exten = 3552,1,Set(TIMEOUT(digit)=2) ; Set Digit Timeout to 5
seconds
exten = 3552,2,Set(TIMEOUT(response)=2) ; Set Response Timeout to
10 seconds
exten = 3552,3,Answer
exten = 3552,4,NoOp(${CALLERID(num)})
exten = 3552,5,Set(number=${CALLERID(num)})
exten = 3552,6,NoOp(${number})
exten = 3552,7,Background(recmsg1) ;Please say yo message after the beep
and end with a hash
exten = 3552,8,Record(crystalrecords/${number}.gsm)
exten = 3552,9,Playback(crystalrecords/${number})
exten = 3552,10,Background(ackrec) ;Press 1 to replay or 2 to
re-record, 3 to save
exten = 3552,11,WaitExten(5)
exten = timo,1,1,Goto,timo|3552|9
exten = timo,2,1,Goto(3552,7) ; re-record message
exten = timo,3,1,Goto(4,1)
exten = timo,4,AGI(timorec.php)
exten = i,1,Background(invalidentry)
exten = i,n,Goto(3552,10)
exten = t,1,Playback(thankyoubye)
exten = t,n,Hangup
In my 1.6 version I use the same configuration in extensions_custom.conf but
I get the error below. It seems like 1.6 does not recognize the button the
user has pressed.
The specific error is
-- Invalid extension '1' in context 'from-internal' on SIP/440-004b
The detailed log is below.
-- Executing [3552@from-internal:1] Set(SIP/440-004b,
TIMEOUT(digit)=2) in new stack
-- Digit timeout set to 2.000
-- Executing [3552@from-internal:2] Set(SIP/440-004b,
TIMEOUT(response)=2) in new stack
-- Response timeout set to 2.000
-- Executing [3552@from-internal:3] Answer(SIP/440-004b, ) in
new stack
-- Executing [3552@from-internal:4] NoOp(SIP/440-004b, 440) in
new stack
-- Executing [3552@from-internal:5] Set(SIP/440-004b,
number=440) in new stack
-- Executing [3552@from-internal:6] NoOp(SIP/440-004b, 440) in
new stack
-- Executing [3552@from-internal:7] BackGround(SIP/440-004b,
recmsg1) in new stack
-- SIP/440-004b Playing 'recmsg1.gsm' (language 'en')
-- Channel 0/2, span 4 got hangup request, cause 16
== Spawn extension (ivr-16, s, 12) exited non-zero on 'DAHDI/95-1'
-- Executing [h@ivr-16:1] Hangup(DAHDI/95-1, ) in new stack
== Spawn extension (ivr-16, h, 1) exited non-zero on 'DAHDI/95-1'
-- Hungup 'DAHDI/95-1'
-- Executing [3552@from-internal:8] Record(SIP/440-004b,
crystalrecords/440.gsm) in new stack
-- SIP/440-004b Playing 'beep.gsm' (language 'en')
-- Executing [3552@from-internal:9] Playback(SIP/440-004b,
crystalrecords/440) in new stack
-- SIP/440-004b Playing 'crystalrecords/440.gsm' (language 'en')
-- Executing [3552@from-internal:10] BackGround(SIP/440-004b,
ackrec) in new stack
-- SIP/440-004b Playing 'ackrec.gsm' (language 'en')
-- Invalid extension '1' in context 'from-internal' on SIP/440-004b
== CDR updated on SIP/440-004b
-- Executing [i@from-internal:1] BackGround(SIP/440-004b,
invalidentry) in new stack
-- SIP/440-004b Playing 'invalidentry.slin' (language 'en')
== Spawn extension (from-internal, i, 1) exited non-zero on
'SIP/440-004b'
-- Executing [h@from-internal:1] Macro(SIP/440-004b, hangupcall)
in new stack
-- Executing [s@macro-hangupcall:1] GotoIf(SIP/440-004b,
1?noautomon) in new stack
-- Goto (macro-hangupcall,s,3)
-- Executing [s@macro-hangupcall:3] NoOp(SIP/440-004b,
TOUCH_MONITOR_OUTPUT=) in new stack
-- Executing [s@macro-hangupcall:4] GotoIf(SIP/440-004b,
1?noautomon2) in new stack
-- Goto (macro-hangupcall,s,6)
-- Executing [s@macro-hangupcall:6] NoOp(SIP/440-004b,
MONITOR_FILENAME=) in new stack
-- Executing [s@macro-hangupcall:7] GotoIf(SIP/440-004b,
1?skiprg) in new stack
-- Goto (macro-hangupcall,s,10)
-- Executing [s@macro-hangupcall:10] GotoIf(SIP/440-004b,
1?skipblkvm) in new stack
-- Goto (macro-hangupcall,s,13)
-- Executing [s@macro-hangupcall:13] GotoIf(SIP/440-004b,
1?theend) in new stack
-- Goto (macro-hangupcall,s,15)
-- Executing [s@macro-hangupcall:15] Hangup(SIP/440-004b, ) in
new stack
== Spawn extension (macro-hangupcall, s, 15) exited non-zero on
'SIP/440-004b' in macro 'hangupcall'
== Spawn extension (from-internal, h, 1) exited non-zero on
'SIP/440-004b'
-- Remote UNIX connection
-- Remote UNIX connection disconnected
Kind Regards
Billy
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Re: [asterisk-users] Custom Application recording problem
Thanks Dale, Am not sure why it was working in 1.4 but for some reason it was ( Note : My Asterisk is running bundled with Elastix). But any your suggestion worked very fine. Now am having one problem how can define those extensions only with in different contexts, the problem I see is since am Building 3 recording applications only one will be able call its AGI file, Say if someone calls custom extension 1114 They can record message -Press 1 to Replay Press 2 to Re-record or Press 3 to Save the file Also if someone calls custom extension 1115 -Press 1 to Replay Press 2 to Re-record or Press 3 to Save the file Note Each save file selection calls a different AGI file E.g exten = 1,1,Goto,timo|3552|9 exten = 2,1,Goto(3552,7) ; re-record message exten = 3,1,Goto(4,1) exten = 4,AGI(timorec.php) Kind Regards Billy On 4/16/12 11:22 PM, Dale Noll dn...@wi.rr.com wrote: On 04/16/2012 08:36 AM, Billy Kaye wrote: In my 1.4 asterisk I have a custom application that users call and make recordings which recording I save to a file with the caller Id. Below is the config file which works perfectly in 1.4 I am not going to say that your application doesn't work under 1.4, but to me it looks like it shouldn't work under 1.4. The issue is that you do not have an extension '1' defined within your context of [timo]. (Not to mention your CLI output appears to be from a different context all together.) When the user presses 1, Asterisk cannot find a valid extension to send the caller to. The reason is these lines are not valid. exten = timo,1,1,Goto,timo|3552|9 exten = timo,2,1,Goto(3552,7) ; re-record message exten = timo,3,1,Goto(4,1) exten = timo,4,AGI(timorec.php) If Asterisk even parses them at all, they would define an extension 'timo' with 4 priorities. I suspect they should be... exten = 1,1,Goto,timo|3552|9 exten = 2,1,Goto(3552,7) ; re-record message exten = 3,1,Goto(4,1) exten = 4,AGI(timorec.php) Dale -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Way to disable codec in dialingplan
Why not have multiple records in the sip.conf for the carrier.
For example, if your carrier was level3 then you'd do something like this:
sip.conf
[level3_729]
host=x.x.x.x
type=peer
insecure=very
context=whatever
disallow=all
allow=g729
[level3_ulaw]
host=x.x.x.x
type=peer
insecure=very
context=whatever
disallow=all
allow=ulaw
In you dialplan, if you wanted to send all calls starting with 407 NPA via
g729 and everything else via ulaw, you'd do the following:
exten = s,1,Ringing
exten = s,2,GoToIf($[${MACRO_EXTEN:-10:3} = 407]?3:5)
exten = s,3,Dial(SIP/[EMAIL PROTECTED])
exten = s,4,hangup
exten = s,5,Dial(SIP/[EMAIL PROTECTED])
exten = s,6,hangup
It may take some tweaking for your needs, but I believe the theory is
sound... it should work.
bp
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Luki
Sent: Thursday, May 25, 2006 9:12 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Way to disable codec in dialingplan
can we enable or force a codec on specified npa..
Depends on the channel. On SIP you can set SIP_CODEC to force a codec,
but I don't think you can disallow one in the dialplan.
See:
http://voip-info.org/tiki-pagehistory.php?page=Asterisk+variables
--Luki
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RE: [Asterisk-Users] OT: AudioCodes MP124-C/FSX/AC/SIP
FXS -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of The VoIP Connection Sent: Wednesday, May 24, 2006 9:25 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] OT: AudioCodes MP124-C/FSX/AC/SIP FXS or FXO? Michael Crown Managing Partner www.thevoipconnection.com 321.989.6728 ext. 611 sip:[EMAIL PROTECTED] -Original Message- From: William Piper [mailto:[EMAIL PROTECTED] Sent: Wednesday, May 24, 2006 6:27 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] OT: AudioCodes MP124-C/FSX/AC/SIP Sorry to hijack your thread. Reading these posts made me grab my old MP-104 try again to get it working with asterisk. I bought it a while ago off eBay never could get it to register. Does anyone have an example ini file for the MP-1XX that I could look at figure out what I am configuring wrong on this box? Thanks, bp ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] plainvoip - IAX2 call rejected
Use this:
exten = _1NXXNXX,2,Dial,IAX2/username:[EMAIL PROTECTED]/${EXTEN}
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joseph
Sent: Sunday, May 14, 2006 2:39 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] plainvoip - IAX2 call rejected
Is anybody using plainvoip provider with IAX2? They seem to support
IAX but it rejects my calls.
-- Executing Dial(SIP/11-cb98, IAX2/[EMAIL PROTECTED]) in new
stack
-- Called [EMAIL PROTECTED]
May 14 00:33:32 WARNING[26580]: chan_iax2.c:5553 socket_read: Call
rejected by 66.199.240.2: No authority found
My registration goes through OK.
My dial plan:
exten = _1NXXNXX,2,Dial,IAX2/[EMAIL PROTECTED]
--
#Joseph
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RE: [Asterisk-Users] ATXFER
Can someone please kill this guy's account? Isn't there a Moderator on this list? bp -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Friday, May 12, 2006 10:05 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] ATXFER Please change the email address of [EMAIL PROTECTED] to [EMAIL PROTECTED] Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ NOD32 1.1535 (20060512) Information __ This message was checked by NOD32 antivirus system. http://www.eset.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SPA-1001 behind NAT -- mucho hair pulling
On a full cone NAT, I have never been able to get the ATA to register without a stun. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Damon Estep Sent: Tuesday, May 02, 2006 1:15 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] SPA-1001 behind NAT -- mucho hair pulling My experience is that a stun server does not do anything that nat=yes in asterisk does not do. Asterisk is capable of determining the source port and ip address of a registration, so there is no need for the UA (ATA) to learn this information form a stun server. Keep it simple if possible, the stun server just adds another device to manage and/or worry about being unreachable. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Monday, May 01, 2006 9:04 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] SPA-1001 behind NAT -- mucho hair pulling You will probably want to set a stun server in the 2100 if behind a nat. You can use stun.fwdnet.net for testing. With that, you probably wont need to port forward it should work. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Damon Estep Sent: Monday, May 01, 2006 8:30 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] SPA-1001 behind NAT -- mucho hair pulling Set nat=yes as you have Enable qualify=yes Important - Do a sip reload or asterisk reload (the nat and qualify settings have to be refreshed, at least with realtime and rtcahcefriends). Turn off all NAT traversal features on the SPA2100 If it still does not work - your NAT router may be the issue, make sure that security policy allows ALL outbound traffic from the SPA2100 (no filters). With Linksys, Belkin, and some 3com/USR NAT routers (among others I am sure) you will need to make sure you have recent firmware on them, older firmware (1 year or older in many cases) does not behave well with SIP and NAT. The NAT=yes tells asterisk to use the IP address and port of the connection socket (a form of NAT discovery similar to a STUN server), not what is in the registration message, and the qualify=yes tells asterisk to send periodic SIP OPTIONS queries to keep the NAT timeout from expiring on the NAT router. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Eric Lyons Sent: Monday, May 01, 2006 5:01 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] SPA-1001 behind NAT -- mucho hair pulling I've got a Sipura SPA-1001 that I'm trying to get working with an Asterisk server that's on the public Internet, while the SPA-1001 is behind NAT. I did the first obvious thing and mapped ports 5060 and 1 - 3 to the local IP address of the SPA-1001. Tried numerous proxy settings, have all the NAT settings == yes. Registration seems to be happening; with sip debug on, I see it get an OK and sip show peers shows it on the list. But I can't get a dial tone. It works fine connecting to a local Asterisk box (not traversing NAT). Anyone know the magic trick? My sip.conf looks like: [homesip] type=friend username=homesip secret=pw context=fagi ;qualify=yes host=dynamic nat=yes tried qualify both ways. My sip show peers says: telebox*CLI sip show peers Name/username HostDyn Nat ACL Port Status homesip/homesip67.188.35.109D N 5060 Unmonitored Can't seem to find enough info to get this to work, any help appreciated greatly, Eric. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ NOD32 1.1447 (20060316) Information __ This message was checked by NOD32 antivirus system. http://www.eset.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ NOD32 1.1447 (20060316) Information __ This message was checked by NOD32 antivirus system. http://www.eset.com ___ --Bandwidth and Colocation provided by
RE: [Asterisk-Users] SPA-1001 behind NAT -- mucho hair pulling
You will probably want to set a stun server in the 2100 if behind a nat. You can use stun.fwdnet.net for testing. With that, you probably wont need to port forward it should work. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Damon Estep Sent: Monday, May 01, 2006 8:30 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] SPA-1001 behind NAT -- mucho hair pulling Set nat=yes as you have Enable qualify=yes Important - Do a sip reload or asterisk reload (the nat and qualify settings have to be refreshed, at least with realtime and rtcahcefriends). Turn off all NAT traversal features on the SPA2100 If it still does not work - your NAT router may be the issue, make sure that security policy allows ALL outbound traffic from the SPA2100 (no filters). With Linksys, Belkin, and some 3com/USR NAT routers (among others I am sure) you will need to make sure you have recent firmware on them, older firmware (1 year or older in many cases) does not behave well with SIP and NAT. The NAT=yes tells asterisk to use the IP address and port of the connection socket (a form of NAT discovery similar to a STUN server), not what is in the registration message, and the qualify=yes tells asterisk to send periodic SIP OPTIONS queries to keep the NAT timeout from expiring on the NAT router. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Eric Lyons Sent: Monday, May 01, 2006 5:01 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] SPA-1001 behind NAT -- mucho hair pulling I've got a Sipura SPA-1001 that I'm trying to get working with an Asterisk server that's on the public Internet, while the SPA-1001 is behind NAT. I did the first obvious thing and mapped ports 5060 and 1 - 3 to the local IP address of the SPA-1001. Tried numerous proxy settings, have all the NAT settings == yes. Registration seems to be happening; with sip debug on, I see it get an OK and sip show peers shows it on the list. But I can't get a dial tone. It works fine connecting to a local Asterisk box (not traversing NAT). Anyone know the magic trick? My sip.conf looks like: [homesip] type=friend username=homesip secret=pw context=fagi ;qualify=yes host=dynamic nat=yes tried qualify both ways. My sip show peers says: telebox*CLI sip show peers Name/username HostDyn Nat ACL Port Status homesip/homesip67.188.35.109D N 5060 Unmonitored Can't seem to find enough info to get this to work, any help appreciated greatly, Eric. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ NOD32 1.1447 (20060316) Information __ This message was checked by NOD32 antivirus system. http://www.eset.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] two box share one real time configuration database.
Im not sure about IAX, but in SIP you can use rtcachefriends=yes in the general section to accomplish this. Dont know about #2 Billy P. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of ?? Sent: Friday, April 28, 2006 8:40 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] two box share one real time configuration database. hi, alll,,, there two asterisk box share one realtime database... and all the client is IAX2.. and registery dynamic... there have some question need to confirm.. 1, when i run iax2 show peers,,,there no show the peers that registed with real time... the same as run iax2 show users..there not show any real time users.. 2, if user1 have registed with box1,, how user2 on box1 and user3 on box2 can find this channel in which box..? because the all channel is registed dynamicly. we can no preconfig it exten= statence... -- Jeffery iaxtel Num: 1-700-576-1311 fwdnet Num: 728150 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Random 1-way audio on IAX2 Connections
Id set your box to DMZ on the router see if the problem exists first. If so, you probably forgot to forward something. Make sure that you forwarded both TCP UDP ports. Billy P. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Aryn Nakaoka Sent: Friday, April 28, 2006 10:10 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Random 1-way audio on IAX2 Connections I have 2 Asterisk servers connected via IAX2 connections. PBX1 is on the internet with a public IP Address - with PRI PBX 2 is behind a NAT router with IAX2 Ports forwarded 1-way audio is an issue with incoming and outgoing calls using the PRI. However whenever 1-way audio occurs, PBX2 can call PBX1 extensions and there are no issues. As well as a restart of asterisk on PBX2 clears up the problem. Any bugs in IAX2? Thanks aryn Aryn H. K. Nakaoka Tri-net Solutions 733 Bishop St. #170 Honolulu, HI 96813 http://www.trinet-hi.com Main : 808.841.1000 Direct: 808.356.2901 Bridge: 808.356.2998 Fax : 808.356.2901 AIM : NTY2K MSN : [EMAIL PROTECTED] sidekick : [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk2Billing
Ive been using it for a few months now. It works great. Needs some documentation but works really good. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Scheda Sent: Monday, April 24, 2006 10:32 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Asterisk2Billing I'm sure this has been asked a million times. Therefore, I must ask again. Generally speaking, what do you guys think of it. It looks pretty good, but for my uses, I'm not sure that a calling card method is the *best* way to go. But, either way, what is the general concensus? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Connecting to a cluster of SIP servers
Although there maybe a better way, this would work: 1. Add the IP's into your sip.conf and set qualify=yes. 2. Make your dialplan something like the following: exten = _X.,1,Dial,SIP/[EMAIL PROTECTED] exten = _X.,2,Hangup exten = _X.,102,Dial,SIP/[EMAIL PROTECTED] exten = _X.,103,Hangup exten = _X.,203,Dial,SIP/[EMAIL PROTECTED] exten = _X.,204,Hangup exten = _X.,304,Dial,SIP/[EMAIL PROTECTED] exten = _X.,305,Hangup This would make your failover work but certainly wouldn't help with the load balancing between the servers. If any cannot qualify or are congested, they will automatically failover to the next server. I believe most people use an SER proxy for this type of application. It seems to work well with the round robin type DNS. William -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Hill Sent: Saturday, April 22, 2006 5:13 AM To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] Connecting to a cluster of SIP servers My Asterisk server is connecting to sip.plus.net, which resolves to multiple IP addresses: sip.plus.net. 300 IN A 84.92.0.75 sip.plus.net. 300 IN A 84.92.0.76 sip.plus.net. 300 IN A 84.92.5.189 sip.plus.net. 300 IN A 84.92.5.190 If one of these machines is down (i.e. it's not replying to the SIP packets or it's sending back ICMP Port Unreachable), Asterisk keeps trying the same server. Shouldn't Asterisk move on to the next server automatically in this case? It seems to only way to do this at the moment is to run the reload command, which causes it to do a DNS lookup and it may then pick one of the other servers. -- - Steve xmpp:[EMAIL PROTECTED] sip:[EMAIL PROTECTED] http://www.nexusuk.org/ Servatis a periculum, servatis a maleficum - Whisper, Evanescence ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ NOD32 1.1447 (20060316) Information __ This message was checked by NOD32 antivirus system. http://www.eset.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] RE: SPA 3000 - UK Replacement
Here is Grandstream's version of the spa-3000. I have used it and it works great with asterisk. http://grandstream.com/y-ht488.htm -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of tom Sent: Saturday, April 22, 2006 8:35 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] RE: SPA 3000 - UK Replacement Steven wrote: First off I am totally annoyed and let down by PC World Business (PCWB part of the Dixons Group). I ordered one of these babies from them over a month ago. After constantly chasing them up they finally told me they couldn't deliver, and have now only just returned the money they stole from me. I only bought from them because they showed a 4-day availability stock level! You think that's bad, I ordered one on the 10th of march from redstore, that was showing a 3-5 day. They still haven't despatched the unit and I have been trying to call them now (on their 0870 number) for about a week, during the past 3 weeks I have been sending them email after email that hasn't been responded to. Now I'm screwed as it seems these are impossible to come by in the UK now since Sipura decided to discontinue it.. Oh Balls! Didn't know that. Broadbandbuyer have them lsited as entering stock on the 25th of May, I'd order one if I knew that I could cancel the redstore one. Now my back to my subject. Does anyone know of a decent replacement for the SPA3000. I need at least 1 FXO and 1 FXS but am willing to pay for 2 FXS on the same unit. What I'm looking for needs to be of a likable price to the SPA3000 which in it's hayday was retailing for around £70 at some outlets. £49.45 + VAT at broadbandbuyer.co.uk or 78 euros from someone on ebay in the netherlands. I'm primarily looking for something network attachable. But could stretch to USB or PCI if the price was right.. I'm steering away from PCI cards as they seem to have terrible issues with UK analogue lines such as not being able to detect hang ups.. (Also; the server I'd ideally like to add this capability too, has no free PCI slots..) I'd be interested in a similar unit myself, if the price is right although I'd prefer a network attachable device myself. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ NOD32 1.1447 (20060316) Information __ This message was checked by NOD32 antivirus system. http://www.eset.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Upgrade from 1.2.4 to 1.2.7.1
List, I wish to upgrade from 1.2.4 to 1.2.7.1 I have downloaded unzipped the file but how do I compile it? Do I need to make clean then make and make upgrade? Or make then make install? Thanks, William ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Upgrade from 1.2.4 to 1.2.7.1
Ok, I thought that was the case but I seem to remember doing make then make upgrade in the past. Is this no longer the way to do it or just another way? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alexander Lopez Sent: Wednesday, April 19, 2006 9:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Upgrade from 1.2.4 to 1.2.7.1 This is the order that is recomended Zaptel libpri asterisk after unzip/untar cd into use directory and run make and then a make install. I would suggest you clean out your modules directory to be safe. rm -rf /usr/lib/asterisk/modules/* From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Wednesday, April 19, 2006 9:47 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Upgrade from 1.2.4 to 1.2.7.1 List, I wish to upgrade from 1.2.4 to 1.2.7.1 I have downloaded unzipped the file but how do I compile it? Do I need to make clean then make and make upgrade? Or make then make install? Thanks, William __ NOD32 1.1447 (20060316) Information __ This message was checked by NOD32 antivirus system. http://www.eset.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Upgrade from 1.2.4 to 1.2.7.1
I deleted the modules directory, then ran make and make install. I then did service asterisk start and asterisk r but it says: [EMAIL PROTECTED] asterisk-1.2.7.1]# asterisk -r Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) [EMAIL PROTECTED] asterisk-1.2.7.1]# From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alexander Lopez Sent: Wednesday, April 19, 2006 10:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Upgrade from 1.2.4 to 1.2.7.1 Sorry forgot about that one. it does a make all and then a bininstall, (installs binaries) It does NOT however remove your modules, I would still do the rm -rf /usr/lib/asterisk/modules/* before I do a make. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Wednesday, April 19, 2006 9:56 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Upgrade from 1.2.4 to 1.2.7.1 Ok, I thought that was the case but I seem to remember doing make then make upgrade in the past. Is this no longer the way to do it or just another way? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alexander Lopez Sent: Wednesday, April 19, 2006 9:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Upgrade from 1.2.4 to 1.2.7.1 This is the order that is recomended Zaptel libpri asterisk after unzip/untar cd into use directory and run make and then a make install. I would suggest you clean out your modules directory to be safe. rm -rf /usr/lib/asterisk/modules/* From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Wednesday, April 19, 2006 9:47 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Upgrade from 1.2.4 to 1.2.7.1 List, I wish to upgrade from 1.2.4 to 1.2.7.1 I have downloaded unzipped the file but how do I compile it? Do I need to make clean then make and make upgrade? Or make then make install? Thanks, William __ NOD32 1.1447 (20060316) Information __ This message was checked by NOD32 antivirus system. http://www.eset.com __ NOD32 1.1447 (20060316) Information __ This message was checked by NOD32 antivirus system. http://www.eset.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Upgrade from 1.2.4 to 1.2.7.1
Thanks, I found it though. I needed the latest addons package. mysql CDR wasnt there. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alexander Lopez Sent: Wednesday, April 19, 2006 11:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Upgrade from 1.2.4 to 1.2.7.1 Run asterisk from the command line and dont put it in the background: like so: asterisk -cvv This will tell you what the error is. fix it and rerun the service. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Wednesday, April 19, 2006 10:49 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Upgrade from 1.2.4 to 1.2.7.1 Importance: High I deleted the modules directory, then ran make and make install. I then did service asterisk start and asterisk r but it says: [EMAIL PROTECTED] asterisk-1.2.7.1]# asterisk -r Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) [EMAIL PROTECTED] asterisk-1.2.7.1]# From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alexander Lopez Sent: Wednesday, April 19, 2006 10:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Upgrade from 1.2.4 to 1.2.7.1 Sorry forgot about that one. it does a make all and then a bininstall, (installs binaries) It does NOT however remove your modules, I would still do the rm -rf /usr/lib/asterisk/modules/* before I do a make. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Wednesday, April 19, 2006 9:56 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Upgrade from 1.2.4 to 1.2.7.1 Ok, I thought that was the case but I seem to remember doing make then make upgrade in the past. Is this no longer the way to do it or just another way? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alexander Lopez Sent: Wednesday, April 19, 2006 9:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Upgrade from 1.2.4 to 1.2.7.1 This is the order that is recomended Zaptel libpri asterisk after unzip/untar cd into use directory and run make and then a make install. I would suggest you clean out your modules directory to be safe. rm -rf /usr/lib/asterisk/modules/* From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Wednesday, April 19, 2006 9:47 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Upgrade from 1.2.4 to 1.2.7.1 List, I wish to upgrade from 1.2.4 to 1.2.7.1 I have downloaded unzipped the file but how do I compile it? Do I need to make clean then make and make upgrade? Or make then make install? Thanks, William __ NOD32 1.1447 (20060316) Information __ This message was checked by NOD32 antivirus system. http://www.eset.com __ NOD32 1.1447 (20060316) Information __ This message was checked by NOD32 antivirus system. http://www.eset.com __ NOD32 1.1447 (20060316) Information __ This message was checked by NOD32 antivirus system. http://www.eset.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk service crashes
Interesting, I haven't set a hostname since I built the server almost a year ago. I wonder why only now would the problem arise. William _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Jones Sent: Wednesday, April 19, 2006 10:56 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Asterisk service crashes I had a similar problem, and it was because my hostname had issues... I'm not sure why/how, but if my hostname was valid, and had a valid fwd/reverse dns entry, everything was OK again.. -Steve _ From: Josué Conti [mailto:[EMAIL PROTECTED] Sent: Wed 4/19/2006 9:19 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk service crashes Hello William. asterisk - g, makes with that you it initiates daemon of asterisk and it is in background. Does not forget in the CLI it to activate the command set verbose X (1-15) to monitor the events in asterisk. I wait to have helped. Greatings 2006/4/19, Steve Totaro [EMAIL PROTECTED]: dump the core i believe -Original Message- From: William Piper [mailto: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ] Sent: Wed 4/19/2006 8:52 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Cc: Subject: RE: [Asterisk-Users] Asterisk service crashes What does asterisk -g do? I'm not finding anything on google. Thanks, William _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Josué Conti Sent: Wednesday, April 19, 2006 7:00 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk service crashes Try asterisk -g Regards Josué 2006/4/19, Gareth Blades [EMAIL PROTECTED]: Enter the 'dmesg' command. It displays a log of kernel messages etc... and may show up a problem. On Wed, 2006-04-19 at 03:03, [EMAIL PROTECTED] wrote: List, The past few days the asterisk service on my server has crashed several times. I have had it running for months and have made no changes to it. When it crashes, I am unable to make calls or gain access to the CLI. The service has been stopped. If I try to start it again (service asterisk start), it will start and run for a few seconds then crash again. After a reboot, it will run successfully for several hours before doing it again. Here is a ps aux of the services while the server is crashed. Does anyone see any service that would have a conflict with the asterisk service? FYI, the only cron I have running is a reboot scheduled once a week. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ NOD32 1.1447 (20060316) Information __ This message was checked by NOD32 antivirus system. http://www.eset.com attachment: winmail.dat___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Don't see my post
First of all, try sending it to the asterisk-biz list. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Rich Sent: Monday, April 17, 2006 10:53 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Don't see my post Hi Folks, I have posted a couple of message to the list and do see them, even after waitin for long time (2 days). Can someone please point me to the rules for posting to this list? I think it had to do with the subject that I was looking for. I was looking for IAX terminiation service that can handle high volumes. Thanks John. Yahoo! Messenger with Voice. Make PC-to-Phone Calls to the US (and 30+ countries) for 2¢/min or less. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk service crashes
List, The past few days the asterisk service on my server has crashed several times. I have had it running for months and have made no changes to it. When it crashes, I am unable to make calls or gain access to the CLI. The service has been stopped. If I try to start it again (service asterisk start), it will start and run for a few seconds then crash again. After a reboot, it will run successfully for several hours before doing it again. Here is a ps aux of the services while the server is crashed. Does anyone see any service that would have a conflict with the asterisk service? FYI, the only cron I have running is a reboot scheduled once a week. USER PID %CPU %MEM VSZ RSS TTY STAT START TIME COMMAND root 1 0.0 0.0 3096 656 ? S 17:39 0:00 init [3] root 2 0.0 0.0 0 0 ? S 17:39 0:00 [migration/0] root 3 0.0 0.0 0 0 ? SN 17:39 0:00 [ksoftirqd/0] root 4 0.0 0.0 0 0 ? S 17:39 0:00 [migration/1] root 5 0.0 0.0 0 0 ? SN 17:39 0:00 [ksoftirqd/1] root 6 0.0 0.0 0 0 ? S 17:39 0:00 [migration/2] root 7 0.0 0.0 0 0 ? SN 17:39 0:00 [ksoftirqd/2] root 8 0.0 0.0 0 0 ? S 17:39 0:00 [migration/3] root 9 0.0 0.0 0 0 ? SN 17:39 0:00 [ksoftirqd/3] root 10 0.0 0.0 0 0 ? S 17:39 0:00 [events/0] root 11 0.0 0.0 0 0 ? S 17:39 0:00 [events/1] root 12 0.0 0.0 0 0 ? S 17:39 0:00 [events/2] root 13 0.0 0.0 0 0 ? S 17:39 0:00 [events/3] root 14 0.0 0.0 0 0 ? S 17:39 0:00 [khelper] root 15 0.0 0.0 0 0 ? S 17:39 0:00 [kacpid] root 47 0.0 0.0 0 0 ? S 17:39 0:00 [kblockd/0] root 48 0.0 0.0 0 0 ? S 17:39 0:00 [kblockd/1] root 49 0.0 0.0 0 0 ? S 17:39 0:00 [kblockd/2] root 50 0.0 0.0 0 0 ? S 17:39 0:00 [kblockd/3] root 60 0.0 0.0 0 0 ? S 17:39 0:00 [pdflush] root 61 0.0 0.0 0 0 ? S 17:39 0:00 [pdflush] root 63 0.0 0.0 0 0 ? S 17:39 0:00 [aio/0] root 64 0.0 0.0 0 0 ? S 17:39 0:00 [aio/1] root 65 0.0 0.0 0 0 ? S 17:39 0:00 [aio/2] root 66 0.0 0.0 0 0 ? S 17:39 0:00 [aio/3] root 51 0.0 0.0 0 0 ? S 17:39 0:00 [khubd] root 62 0.0 0.0 0 0 ? S 17:39 0:00 [kswapd0] root 139 0.0 0.0 0 0 ? S 17:39 0:00 [kseriod] root 204 0.0 0.0 0 0 ? S 17:39 0:00 [scsi_eh_0] root 205 0.0 0.0 0 0 ? S 17:39 0:00 [aacraid] root 217 0.0 0.0 0 0 ? S 17:39 0:00 [kmirrord] root 218 0.0 0.0 0 0 ? S 17:39 0:00 [kmir_mon] root 226 0.0 0.0 0 0 ? S 17:39 0:00 [kjournald] root 1092 0.0 0.0 2312 556 ? Ss 17:39 0:00 udevd root 1125 0.0 0.0 0 0 ? S 17:39 0:00 [shpchpd_event] root 1377 0.0 0.0 0 0 ? S 17:39 0:00 [kauditd] root 1424 0.0 0.0 0 0 ? S 17:39 0:00 [kjournald] root 1962 0.0 0.0 2612 668 ? Ss 17:40 0:00 syslogd -m 0 root 1966 0.0 0.0 3532 532 ? Ss 17:40 0:00 klogd -x root 1976 0.0 0.0 2128 540 ? Ss 17:40 0:00 irqbalance rpc 1993 0.0 0.0 3432 644 ? Ss 17:40 0:00 portmap root 2012 0.0 0.0 1800 912 ? Ss 17:40 0:00 rpc.statd root 2038 0.0 0.0 5156 1140 ? Ss 17:40 0:00 rpc.idmapd root 2115 0.0 0.0 2092 648 ? Ss 17:40 0:00 /usr/sbin/acpid root 2124 0.0 0.1 8264 2188 ? Ss 17:40 0:00 cupsd root 2165 0.0 0.0 5120 1724 ? Ss 17:40 0:00 /usr/sbin/sshd root 2188 0.0 0.0 3152 896 ? Ss 17:40 0:00 xinetd -stayalive root 2295 0.0 0.1 9300 3084 ? Ss 17:40 0:00 sendmail: accepti smmsp 2303 0.0 0.1 7252 2668 ? Ss 17:40 0:00 sendmail: Queue r root 2316 0.0 0.0 2984 620 ? Ss 17:40 0:00 gpm -m /dev/input root 2329 0.0 0.0 5412 1176 ? Ss 17:40 0:00 crond xfs 2355 0.0 0.0 3392 1540 ? Ss 17:40 0:00 xfs -droppriv -da root 2373 0.0 0.0 3700 788 ? Ss 17:40 0:00 /usr/sbin/atd dbus 2382 0.0 0.0 3660 1300 ? Ss 17:40 0:00 dbus-daemon-1 --s root 2391 0.0 0.2 8456 5620 ? Ss 17:40 0:00 hald root 2471 0.0 0.2 16544 6000 ? Ss 17:40 0:00 /usr/sbin/httpd - root 2483 0.0 0.0 4700 1068 ? S 17:40 0:00 /usr/sbin/vsftpd root 2555 0.0 0.0 4544 1304 ? S 17:40 0:00 /bin/sh /usr/bin/ asterisk 2575 0.0 0.2 16544 6124 ? S 17:40 0:00 /usr/sbin/httpd - asterisk 2578 0.0 0.2 16544 6128 ? S 17:40 0:00 /usr/sbin/httpd - asterisk 2579 0.0 0.2 16544 6128 ? S 17:40 0:00 /usr/sbin/httpd - asterisk 2581 0.0 0.2 16544 6128 ? S 17:40 0:00 /usr/sbin/httpd - asterisk 2583 0.0 0.2 16544 6128 ? S 17:40 0:00 /usr/sbin/httpd - asterisk 2585 0.0 0.2 16544 6128 ? S 17:40 0:00 /usr/sbin/httpd - asterisk 2586 0.0 0.2 16544 6128 ? S 17:40 0:00 /usr/sbin/httpd - asterisk 2588 0.0 0.2 16544 6128 ? S 17:40 0:00 /usr/sbin/httpd - mysql 2600 0.0 0.9 125808 19424 ? Sl 17:40 0:00 /usr/libexec/mysq root 2697 0.0 0.0 1856 508 tty1 Ss+ 17:40 0:00 /sbin/mingetty tt root 2698 0.0 0.0 2600 508 tty2 Ss+ 17:40 0:00 /sbin/mingetty tt root 2699 0.0 0.0 1856 508 tty3 Ss+ 17:40 0:00 /sbin/mingetty tt root 2700 0.0 0.0 3184 508 tty4 Ss+ 17:40 0:00 /sbin/mingetty tt root 2701 0.0 0.0 2352 516 tty5 Ss+ 17:40 0:00 /sbin/mingetty tt root 2702 0.0 0.0 2488 508 tty6 Ss+ 17:40 0:00 /sbin/mingetty tt root 3754 0.0 0.1 6984 2244 ? Ss 17:56 0:00 sshd: bpiper [pri bpiper 3768 0.0 0.1 6984 2320 ? S 17:56 0:00 sshd: [EMAIL PROTECTED]/ bpiper 3769 0.0 0.0 4560 1456 pts/1 Ss 17:56 0:00 -bash root 3795 0.0 0.0 5696 1240 pts/1 S 17:56 0:00 su root 3796 0.0 0.0 4608 1464 pts/1 S+
[Asterisk-Users] Slow outgoing pstn calls
Hi.. Have AAH set up with tdm card. 1 pstn line. When incoming call initiated hard phone rings almost instantly. Problem with outgoing calls from sipura spa 941, the call connects etc, but is very slow to go out onto pstn. There is a significant lag before the call at other end rings, perhaps as much as 7 seconds Is there any way shorten this ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] low call volume
i have AAH connected to pstn via digium TDM01B had been testing it on telewest line (UK cable company) with very little issues.now moved to a BT line and had several that i anticipated from infomation on this list.the one that has caught me out is low volume from the caller via pstn. using sipura spa-941's and have to push the volume up to hear.is there a setting that can correct this thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] trunk not registering -newbie
Hello: My by-the-book [EMAIL PROTECTED]/Broadvoice installation doesn't register in the SIP Registry, IAX2 SIP Registry, or with SIP peers. The Asterisk server is behind a firewall using NAT. The checkpoint firewall opens up all IP Telephony ports and I manually opened up ports 4000-2. There do not seem to be any issues related to the NAT, firewall or network. I tried re-directing port 5060 to the asterisk server and adding this to the sip_additional.conf: port = 5060 externIP = 62.219.212.2 localnet = 192.168.10.0 localmask 255.255.255.0 nat=1 I got the same results when I set the trunk up as IAX2. This is what show up in the Asterisk info: Sip Registry Name/usernameHostDyn Nat ACL Mask Port Status Verbosity is at least 3 -- Remote UNIX connection disconnected Sip Peers HostUsername Refresh State -- Remote UNIX connection IAX2 Sip Registry Host UsernamePerceived Refresh State -- Remote UNIX connection -- Remote UNIX connection disconnected IAX2 Peers Name/UsernameHost Mask Port Status bv/3109439023147.135.20.128 (S) 255.255.255.255 4569 Unmonitored -- Remote UNIX connection -- Remote UNIX connection disconnected These are the errors that appear in the log file: Nov 26 06:22:54 DEBUG[2018]: Unable to find key 'IAX2/BV' in family 'cfb' Nov 26 06:22:54 DEBUG[2018]: Manager received command 'Command' Nov 26 06:22:54 DEBUG[2018]: Unable to find key 'IAX2/SIP.BROADVOICE.COM' in family 'cfb' Nov 26 06:22:54 DEBUG[2018]: Manager received command 'Command' Nov 26 06:22:54 DEBUG[2018]: Manager received command 'Command' Nov 26 06:22:54 DEBUG[2018]: Unable to find key 'SIP/1000' in family 'cfb' Nov 26 06:22:54 DEBUG[2018]: Manager received command 'Command' Nov 26 06:22:54 DEBUG[2018]: Unable to find key 'IAX2/BV' in family 'dnd' Also: Nov 25 11:51:10 WARNING[2019]: mybvpassword is not a valid port number at line 1 Also: Nov 26 06:22:54 WARNING[2018]: Unknown directive 'permit=192.168.1.0/255.255.255.0' at line 18 of manager_custom.conf even though this is rem'ed out in manager_custom.conf #permit=192.168.1.0/255.255.255.0 I followed the instruction listed at: http://mundy.org/blog/index.php?p=66 this is my sip_additional.conf: [EMAIL PROTECTED]:mybvpassword:[EMAIL PROTECTED] [bv] username=3109439023 user=phone type=peer secret=mybvpassword nat=yes insecure=very host=sip.broadvoice.com fromuser=3109439023 fromdomain=sip.broadvoice.com dtmfmode=inband dtmf=inband canreinvite=no authname=3109439023 [sip.broadvoice.com] username=3109439023 user=3109439023 type=user secret=mybvpassword nat=yes insecure=very host=sip.broadvoice.com fromdomain=sip.broadvoice.com dtmfmode=rfc2833 dtmf=rfc2833 context=from-pstn Any help would be appreciated. Thanks, Billy Troper __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dual PRI fail over
NFAS isn't needed for redundancy.. You can do this with 2 PRI's just have them in the same huntgroup at the CO.. All NFAS really does is free up a extra B Channel.. For 2 PRI's it makes no sense.. For anything over 2 you pickup 1 extra B channel per PRI.. We use NFAS on our RAS setup for the ISP.. We get a extra 6 lines on our 8 PRI group. - Original Message - From: Kevin P. Fleming [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Wednesday, October 12, 2005 12:48 AM Subject: Re: [Asterisk-Users] Dual PRI fail over Tom wrote: I currently have a single PRI however we are getting a second PRI, and the provider (qwest) wants to know if our PBX supports GSAS (they say its a redundant d-channel technology but searching on google for GSAS reveals less than nothing). I've set something similar up before on a cisco 5350, where if one of the PRIs fails, all of the calls destined for either PRI will be routed down the one that didn't fail. Basically the 2 PRIs are bonded together, and act as one. During normal operation the calls come down each PRI in a load balanced fashion (IE if I've got 30 calls up, 15 will be on one PRI and 15 on the other). The term you are looking for is NFAS (Non-Facilities Associated Signaling), and it's fully supported in Asterisk. You can configure your two PRIs as a single trunk group with a primary and backup D-channel, and calls can be handled on both PRIs equally. ___ --Bandwidth and Colocation sponsored by Easynews.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 ___ --Bandwidth and Colocation sponsored by Easynews.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
Re: [Asterisk-Users] Dual PRI fail over
See this link: http://www.voip-info.org/tiki-index.php?page=NFAS - Original Message - From: Billy Huddleston [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, October 12, 2005 12:57 AM Subject: Re: [Asterisk-Users] Dual PRI fail over NFAS isn't needed for redundancy.. You can do this with 2 PRI's just have them in the same huntgroup at the CO.. All NFAS really does is free up a extra B Channel.. For 2 PRI's it makes no sense.. For anything over 2 you pickup 1 extra B channel per PRI.. We use NFAS on our RAS setup for the ISP.. We get a extra 6 lines on our 8 PRI group. - Original Message - From: Kevin P. Fleming [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Wednesday, October 12, 2005 12:48 AM Subject: Re: [Asterisk-Users] Dual PRI fail over Tom wrote: I currently have a single PRI however we are getting a second PRI, and the provider (qwest) wants to know if our PBX supports GSAS (they say its a redundant d-channel technology but searching on google for GSAS reveals less than nothing). I've set something similar up before on a cisco 5350, where if one of the PRIs fails, all of the calls destined for either PRI will be routed down the one that didn't fail. Basically the 2 PRIs are bonded together, and act as one. During normal operation the calls come down each PRI in a load balanced fashion (IE if I've got 30 calls up, 15 will be on one PRI and 15 on the other). The term you are looking for is NFAS (Non-Facilities Associated Signaling), and it's fully supported in Asterisk. You can configure your two PRIs as a single trunk group with a primary and backup D-channel, and calls can be handled on both PRIs equally. ___ --Bandwidth and Colocation sponsored by Easynews.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 ___ --Bandwidth and Colocation sponsored by Easynews.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 ___ --Bandwidth and Colocation sponsored by Easynews.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
[Asterisk-Users] realtime sip channel configuration - insecure option
Hi all
I'm trying to figure out what values are valid for the insecure option in a
realtime configuration table. The table field is 4 chars long and the actual
valid values for this is longer. Can I modify the field length or has this
changed? Below is where I looked, if I'm not looking in the right place
please let me know.
the field on the table is:
...
`insecure` varchar(4) default NULL,
...
(http://www.voip-info.org/tiki-index.php?page=Asterisk+RealTime+Sip)
the actual values for this option (that I have found) are:
port: ignore the port number where authentication came from
invite: don't require initial INVITE to authenticate
port,invite: don't require initial INVITE to authenticate and ignore the port
where the request came from
(http://www.voip-info.org/tiki-index.php?page=Asterisk+sip+insecure)
also found this on chan_sip.c:
/*--- insecure2str: Convert Insecure setting to printable string ---*/
static const char *insecure2str(int port, int invite)
{
if (port invite)
return port,invite;
else if (port)
return port;
else if (invite)
return invite;
else
return no;
}
thanks
Guillermo Krepper
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[Asterisk-Users] Polycom 600 Presence indications - ALWAYS OFF-HOOK?
I am having one problem with the Polycom 600 phones. All phones on the local network are fine and indicate presence to other phones perfectly. One phone that is outside the network can see presence indications of the other phones correctly, but that phone always shows as off the hook to certain phones on the network. Any ideas? Thanks! ___ 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] Polycom 600 Presence indications - ALWAYS OFF-HOOK? - SOLVED
Billy Dunn wrote: I am having one problem with the Polycom 600 phones. All phones on the local network are fine and indicate presence to other phones perfectly. One phone that is outside the network can see presence indications of the other phones correctly, but that phone always shows as off the hook to certain phones on the network. Any ideas? Thanks! ___ 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 Nevermind. We resolved the issues by loading new firmware and/or rebooting those certain phones. ___ 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] Automatic setup of calls between two external lines
Rob Scott wrote: Is it possible to automatically set up a call between two external lines? I would like Asterisk is call a cellphone number, wait for it to answer, and then call another cellphone, when that answers connect the two together. I assume it is possible but can someone point me how to do it. Thanks. ___ 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 I think this is possible, but the conversation will look like this: 1st cell phone -- asterisk -- 2nd cell phone ___ 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] mpg123 - two processes
Does everyone have two processes running for mpg123? I always have them when I'm running an idle Asterisk box. No calls going in or out and nothing off hook. Is this normal? Thanks! 5008 ?S 0:00 mpg123 -q -s --mono -r 8000 -b 2048 -f 4096 fpm-calm-ri 5015 ?S 0:00 /usr/sbin/asterisk 5061 ?S 0:00 mpg123 -q -s --mono -r 8000 -b 2048 -f 4096 fpm-calm-ri ___ 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: Polycom 600 one-touch message access?
Louis-David Mitterrand wrote:
On Mon, Jul 25, 2005 at 09:38:29AM -0400, Noah Miller wrote:
With the 1.5.2 firmware, have you managed to get one-touch message access
when
pressing the "Messages" button? It worked for me with 1.4.1 but no longer
with
1.5.2: I have to go through the message count screen first.
In phone.cfg I have:
msg msg.bypassInstantMessage="1"
In the phone.cfg file under the above line, make sure you also have:
mwi msg.mwi.1.subscribe="" msg.mwi.1.callBackMode="contact" msg.mwi.
1.callBack="Your-VM-Exten" ...
Yes, I have that setup too (no change from 1.4.1)
Are you saying one-touch voicemail works for you with 1.5.2 ?
(meaning no message count summary screen when pressing "Messages")
This is how I did it and it works like a charm:
In phone.cfg:
msg msg.bypassInstantMessage="1"
mwi msg.mwi.1.subscribe="" msg.mwi.1.callBackMode="contact"
msg.mwi.1.callBack="4000"/
/msg
In extensions.conf:
exten = 4000,1,Answer
exten = 4000,2,VoicemailMain(s${CALLERIDNUM})
exten = 4000,3,Wait(1)
exten = 4000,4,Hangup
Obviously it doesn't have to be extension 4000... I just picked that at
random.
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Re: [Asterisk-Users] Polycom IP600 - Flashing clock and date?
[EMAIL PROTECTED] wrote: There should be a NTP setting. Setup Network Time Protocol. ___ 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 This was a pain in the butt for me. In fact, I only was able to get it going by pointing the SNTP server to pool.ntp.org and making sure the DNS entries were correct. That works, but it's not a great solution. When the phone is flashing, that means it cannot contact the SNTP servers. Ideally it should talk to a local NTP server on your network, but I have yet to see that work (but I'm only two weeks into Asterisk too). Good luck. ___ 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] Polycom IP600 - Flashing clock and date?
Chris Mason (Lists) wrote: Ideally it should talk to a local NTP server on your network, but I have yet to see that work (but I'm only two weeks into Asterisk too). Good luck. Works for me, let me know if you need configs Yes, I could use the configs on this. The phones are syncing to pool.ntp.org just fine, but when I point them to my internal NTP server, it fails every time. Thanks. ___ 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] Polycom 600 Ring-Answer (but not ring!)
I have a bunch of Polycom Soundpoint 600 phones and they are working
great. The only thing I can't seem to get them to do is to ring-answer
without the ring.
This is what I have in my sip.cfg file on the boot server:
alertInfo voIpProt.SIP.alertInfo.2.value=RA
voIpProt.SIP.alertInfo.2.class=4/
alertInfo voIpProt.SIP.alertInfo.3.value=RANR
voIpProt.SIP.alertInfo.3.class=5/
RING_ANSWER se.rt.4.name=Ring Answer se.rt.4.type=ring-answer
se.rt.4.timeout=2000 se.rt.
4.ringer=2 se.rt.4.callWait=6 se.rt.4.mod=1/
RING_ANSWER_NR se.rt.5.name=Ring Answer NR se.rt.5.type=answer
se.rt.5.timeout=1 se.rt.5.
ringer=1 se.rt.5.callWait=6 se.rt.5.mod=1/
This is what I have in my extensions.conf file:
; bdunn's Office Extension AUTO ANSWER - WITH RING
exten = 83004,1,SetVar(ALERT_INFO=RA)
exten = 83004,2,Macro(stdexten,3004,${BDUNNOFFICE})
; bdunn's Office Extension AUTO ANSWER - NO RING
exten = 93004,1,SetVar(ALERT_INFO=RANR)
exten = 93004,2,Macro(stdexten,3004,${BDUNNOFFICE})
This does most of what I need - 93004 answers to speakerphone
automatically, but there is a ring (a very short ring). Dialing 83004
gives a moderate length ring and answers as expected. I'd really like
to do it without the ring if possible.
If there are any Polycom pros out there, I could use some help. I have
already checked out this:
http://www.voip-info.org/tiki-index.php?page=Polycom%20auto-answer%20config#comments
Thanks.
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Re: [Asterisk-Users] Polycom 600 Ring-Answer (but not ring!)
I'll give you an example. My staff does a lot of interviews with new hires and salesmen. I do not personally sit in on the whole thing (they can really drag out!), but I would like to be able to listen in to make notes as needed while doing other work. I had another idea whereas I could dial a certain extension which would connect me to all available Polycoms and allow me to announce something. The ring is less important here. Mostly I'm at the point where I think it should be working, but it isn't, and it's driving me a little nuts. :-) Thanks! [EMAIL PROTECTED] wrote: Give me an idea of your application. I personally created a really cool asterisk system for the US Army where the phones would ring on silent for about an hour before people would pick up for a conference call. I Know this config like the back of my head. Keep in mind, Asterisk has many really cool back doors! Tell me your application Brad ___ 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
Re: [Asterisk-Users] Polycom 600 Ring-Answer (but not ring!)
dbruce wrote: If you use the polycom provided config files, the default ring_answer class is 4 and the auto_answer class is 3. So for your RANR alertinfo entry, change the class to 3 and it will work as you expect. ie: alertInfo voIpProt.SIP.alertInfo.3.value=RANR voIpProt.SIP.alertInfo.3.class=3/ The correct ringtype entry should be: AUTO_ANSWER se.rt.3.name=Auto Answer se.rt.3.type=answer se.rt.3.mod=0 / Notice there is no timeout, ringer or callwait entry. Regards, Derek I think that's where I went wrong. I didn't see anything like that in the admin guide, but I'll look again. Thanks very much! ___ 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] Polycom 600 Ring-Answer (but not ring!)
Kristian Kielhofner wrote:
Billy Dunn wrote:
I have a bunch of Polycom Soundpoint 600 phones and they are working
great. The only thing I can't seem to get them to do is to
ring-answer without the ring.
This is what I have in my sip.cfg file on the boot server:
alertInfo voIpProt.SIP.alertInfo.2.value=RA
voIpProt.SIP.alertInfo.2.class=4/
alertInfo voIpProt.SIP.alertInfo.3.value=RANR
voIpProt.SIP.alertInfo.3.class=5/
RING_ANSWER se.rt.4.name=Ring Answer se.rt.4.type=ring-answer
se.rt.4.timeout=2000 se.rt.
4.ringer=2 se.rt.4.callWait=6 se.rt.4.mod=1/
RING_ANSWER_NR se.rt.5.name=Ring Answer NR se.rt.5.type=answer
se.rt.5.timeout=1 se.rt.5.
ringer=1 se.rt.5.callWait=6 se.rt.5.mod=1/
This is what I have in my extensions.conf file:
; bdunn's Office Extension AUTO ANSWER - WITH RING
exten = 83004,1,SetVar(ALERT_INFO=RA)
exten = 83004,2,Macro(stdexten,3004,${BDUNNOFFICE})
; bdunn's Office Extension AUTO ANSWER - NO RING
exten = 93004,1,SetVar(ALERT_INFO=RANR)
exten = 93004,2,Macro(stdexten,3004,${BDUNNOFFICE})
This does most of what I need - 93004 answers to speakerphone
automatically, but there is a ring (a very short ring). Dialing
83004 gives a moderate length ring and answers as expected. I'd
really like to do it without the ring if possible.
If there are any Polycom pros out there, I could use some help. I
have already checked out this:
http://www.voip-info.org/tiki-index.php?page=Polycom%20auto-answer%20config#comments
Thanks.
My Polycom configs at:
http://www.krisk.org/asterisk/pcom/
will do what you are looking for. Set your ALERT_INFO variable to
equal AA (Auto Answer) and the phone will go right to speaker phone
mode, no ring.
Oh yeah... I've been ripping off your work a lot. It's been a big help
to me. Thanks. I think the next message clears up where I went wrong.
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[Asterisk-Users] Asterisk to Avaya PBX using TDM cards
Hi I'm new in this field, have been reading a lot, and have a little question. could it be possible to connect an Avaya IP office pbx to asterisk using a E1/T1/Pri? Original instalation: Telefone company|Pri---Pri|IP Pffice My Question: Telefone company|Pri ---TDM|Asterisk|TDM ---Pri|IP Office I know that it can be done by using h323, but I need a card on the IPOffice my problem is that I have no more room for expantion on this pbx so I was thinking instead of upgrading the IPOffice maybe I can start using *. The secret of success is converting your problems into opportunities Thanks to everyone. Billy ___ 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] Over 10,000 lines. Will asterisk manage?
A SIP phone *could* normally send its media stream directly from phone to phone, if no transcoding is required, but when using Asterisk the media stream will always pass through the server, causing a pottential bottleneck. So, why not use SER to register all the SIP phones, as it doesn't handle the media-streams, just keeps track of the phones and does the 'handshake'. SER is supposed to be able to handle over 50.000 calls at a time, so one SER server would be enough. Then interface this with one (or more) Asterisk servers to connect to the local PSTN. But maybe I'm missing something fundamental, in which case I'm happy to learn. Um, Wrong, You can do re-invites and have the media go point-to-point, We do it all the time. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Cisco 7960 + 7914 - not worked
7914 isn't supported with asterisk as of yet. - Original Message - From: Vasiliy Voropaev To: [EMAIL PROTECTED] Sent: Friday, October 15, 2004 6:32 PM Subject: [Asterisk-Users] Cisco 7960 + 7914 - not worked I have Cisco 7960 with 7914 operator console. 7960 successfully registered and working with chan_sccp2, but the buttons on the 7914 are all red. What may be wrong? sccp.conf: [SEP] description = VVG type = 7914 context = sip autologin = 821 speeddial = 11,Test1 speeddial = 12,Test2 Firmware version 3.1(MF.G2) "Expansion Module Stats" menu displays that Link State is "Not Supported". May be i need to upgrade firmware? Or add some extra option in SEP.cnf.xml ? Sorry for bad English. Best Regards, Vasiliy Voropaev. ___Asterisk-Users mailing list[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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: Polycom SIP 1.3.1 Reject Button
Sounds like you need to talk to polycom about a reduction in the capabilities of thier phone after the upgrade and have them move the menu option back.. - Original Message - From: Tor Setane [EMAIL PROTECTED] To: [EMAIL PROTECTED] Cc: [EMAIL PROTECTED] Sent: Thursday, September 09, 2004 8:57 AM Subject: [Asterisk-Users] RE: Polycom SIP 1.3.1 Reject Button Brent D. Franks wrote: Hello, I recently upgraded to Sip 1.3.1 and noticed that the Reject Button is no longer appearent on the screen when a second incoming call comes in unless I press the hold button on the first call. Does anyone have a work around for this to reject a call while continuing to talk to the first party? I should also point out that I don't want it to be on *, as the situation varies from call to call. E.g. setting a count limit on a phone is not acceptable, as if the secretary is talking to someone from home, she can put them on hold and take the second call. Thanks, Brent D. Franks I can only answer for the IP600 - when I want to reject the second incoming call, I can do that with the Do Not Disturb button, or I can press blue down arrow and than use the reject soft key. The ongoing call is not interrupted and I don't have to put it on hold first. Regards, Tor Setane ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 GW and DTMF problems
I'm stuck running a old copy of asterisk because something strange is going on in later versions of the CVS.. When I call in from a PSTN into my cisco 2610XM gateway which then routes the call to my asterisk box via sip.. Asterisk can no longer process DTMF tones generated by the calling party. This affects DISA, prompts and menus.. Has anyone else had this problem?? and use.. I DO have dtmf-relay rtp-nte toggled in my dial peer.. Thanks, Billy +--+ | Billy Huddleston Senior Systems Administrator | | Net-Express http://www.nxs.net | | 114 Sherway Rd. Voice: 865-691-2011 | | Knoxville, TN 37922 Fax: 865-691-9894 | | [EMAIL PROTECTED]| +--+ ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Cisco GW and DTMF problems
c2600-is5-mz.123-9 rfc2833 - Original Message - From: Tenorio, Leandro [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Wednesday, September 08, 2004 10:37 PM Subject: RE: [Asterisk-Users] Cisco GW and DTMF problems What version of IOS 're u using, and what's your dtmfmode in *? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Billy Huddleston Sent: Wednesday, September 08, 2004 6:29 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Cisco GW and DTMF problems I'm stuck running a old copy of asterisk because something strange is going on in later versions of the CVS.. When I call in from a PSTN into my cisco 2610XM gateway which then routes the call to my asterisk box via sip.. Asterisk can no longer process DTMF tones generated by the calling party. This affects DISA, prompts and menus.. Has anyone else had this problem?? and use.. I DO have dtmf-relay rtp-nte toggled in my dial peer.. Thanks, Billy +--+ | Billy Huddleston Senior Systems Administrator | | Net-Express http://www.nxs.net | | 114 Sherway Rd. Voice: 865-691-2011 | | Knoxville, TN 37922 Fax: 865-691-9894 | | [EMAIL PROTECTED]| +--+ ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] Asterisk GUIs at Astricon * REMINDER *
Dude, don't flam.. People don't use HTML capable E-Mail programs, or turn off html for reason.. Like spam and web-bugs, and/or using classic email programs like pine and mutt and linux. Geez.. - Original Message - From: Karl J. Vesterling To: [EMAIL PROTECTED] Sent: Sunday, August 01, 2004 10:17 PM Subject: RE: [Asterisk-Users] Asterisk GUIs at Astricon * REMINDER * Get an HTML capable E-Mail Program. They've been freely available for nearly 10 years now and on just about every platform that's got more than a 16 bit bus. PS: How do you manage/configure asterisk from punch card / paper tape? Just curious, I thought you might know... At 07:37 PM 8/1/2004, you wrote: 1. Don't post in html...I have to scroll three pages to the right just to read it. Best Regards, Karl J. Vesterling E-Mail: [EMAIL PROTECTED] Yahoo Messenger: karl_vesterling ICQ: 1548052 AOL Instant Messenger: n2vqm Telephone: Washington DC: (202) 448-3009 Extension 0 Annapolis MD: (240) 524-6706 Extension 0 Seattle WA: (360) 516-1822 Extension 0 Niagara Falls NY: (716) 286-9175 Extension 0 Buffalo NY: (716) 608-1121 Extension 0 United Kingdom: 0870 3403428 Extension 0 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] oh323 codec G7231A6K3
Asterisk doesn't support any form of G723 except with Pass through... You might try G726 or G729 Thanks, Billy - Original Message - From: Arnaud Pignard [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, July 23, 2004 12:45 PM Subject: [Asterisk-Users] oh323 codec G7231A6K3 Hi, I would like use codec G7231A6K3 with oh323, but seems asterisk don't undestood this codec. I can't use G7231, the remote gateway don't accept this version of G723. Thanks for help. -- Arnaud Pignard ([EMAIL PROTECTED]) Frontier Online - Opérateur Internet ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] BRI dead in USA?
TN is one of the BEST states for BRI's.. Bellsouth messed up and had to make some concessions to the PUC a long time ago.. You can get BRI anywhere, and it's a flat fee.. typically $80-$90 per month Biz rate, $35-45 Residential.. Thanks, Billy +--+ | Billy HuddlestonSenior System Administrator | | Net-Express http://www.nxs.net | | 114 Sherway Rd. Voice: 865-691-2011 | | Knoxville, TN 37922 Fax: 865-691-9894 | | [EMAIL PROTECTED]| +--+ - Original Message - From: Steven Critchfield [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, July 20, 2004 5:27 PM Subject: RE: [Asterisk-Users] BRI dead in USA? On Tue, 2004-07-20 at 14:15, Scott Stingel wrote: Brian- Wow - that is high! I got quoted only $35/month for BRI (and a hefty installation) - not too bad. But the comment about no CLI scared me off. In Nashville, it is $90/month for a BRI without per minute charges for business. I remember paying $35/month for residential about 10 years or so ago. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of brian Sent: Tuesday, July 20, 2004 9:46 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] PRI dead in USA? Well they fail to realize that ISDN is used for more than data. I just wanna scream at them and say IT DOES VOICE TO YOU NINNY!.. Rates are far from reasonable. 167/mth here is what I would have to pay for ISDN-BRI. SBC is lame. bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Scott Stingel Sent: Tuesday, July 20, 2004 10:37 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] PRI dead in USA? Hi- Because a majority of my customers are in Europe, I've gotten quite used to working with ISDN (PRI) and BRI on a regular basis. Recently one of my customers asked me if I could terminate a few lines locally here in the USA (California), so I called up SBC to enquire as to how much it would cost to install a BRI here. Although the rates were reasonable (except the installation), I got the distinct impression that they really didn't want to install BRI's. Their comments were well, BRI is getting quite antiquated, and the like. They said with the advent of ADSL, there's not much of a market anymore, as most of past usage was modem related. I'm a little worried about the pricing going up, and availability going down in the near future. I don't have the volume yet to justify PRI. What are other's experience in the US with BRI? Also, they mentioned that I couldn't get caller ID with the BRI service, which I thought was a built- in feature. Thanks Scott Stingel Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] T1 Hardware Echo Can
Do these work with PRI's as well? What's a ball park price on these? Thanks, Billy - Original Message - From: Robb Meredeth [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, July 09, 2004 8:53 PM Subject: Re: [Asterisk-Users] T1 Hardware Echo Can Well, first off we're not using them directly off the asterisk. Our Asterisk runs into our Alcatel OmniPCX 4400 and then to the PRI. We have about 70 or so IP phones off the 4400 and just a few off the Asterisk box (so far) and of those probably 10 to 15 (alcatel) users complain consistently about the echo. We've only put it on one of our spans to start with and I directed the most complaining users to it. So far the feedback has been very good. I use an IP phone every day and I rarely notice it, but I may have just learned to tune it out. :) But having said that I have noticed that even that occasional echoing call that I was getting seems to have gone away. Robb - Original Message - From: Brent Franks [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, July 09, 2004 7:41 PM Subject: RE: [Asterisk-Users] T1 Hardware Echo Can Ditech Communications ( http://www.ditechcom.com/ ) has a 2 slot and a 4 slot chassis and you can populate them with just 1 echo cancelling card if you like. They have RJ45 jacks and you have to use a T1 Cross cable on both sides. We Hi Robb, Thanks for your reply. Have you found that the Echo Can's from Ditech take care of your echo? Do you experience echo on any calls (e.g. 1 out of 300 calls, 1 out of 500..?) Thanks a lot for your advice on this. I will give them a call. Additionally, what is a ball park range on pricing? Thanks in advance! - Brent ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] asterisk grandstream aleatory error
I've been working with Mark today on fixing this very bug.. The patch ProgramerTED did may have fixed it, but, I don't think it was the right fix. We should have something done later today on this problem. Thanks, Billy aka Connor - Original Message - From: Alberto Fernandez [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, July 07, 2004 10:37 AM Subject: Re: [Asterisk-Users] asterisk grandstream aleatory error There is an open BUG for this problem, ProgramerTED sent also a fix for this issue. But it still not in the CVS, Im waiting with a verry OLD version because of it. Roll back to before 6/15/02 and you will be fine. Hopefully soon someone can include that fix into the cvs. On Tue, 2004-07-06 at 20:55, interopen wrote: Hello all, It just start happening a week ago on a handytone 286, sip extension, every time i call to this extension it rings one time and it hungs, in the asterisk console this error repeats: Jul 7 00:19:32 WARNING[98311]: chan_sip.c:471 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Request) I change the ip of the handytone and work for a couple of days but start again. Any solution or trick. Thanks in advanced, Ivan ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] Echo cancellation, when software doesn't cut it. Whats next?
I've got the same problem NEAR end echo (We hear the echo on OUR side, person on the PSTN never hears it..) We're tyring to get our PRI carrier to run us through a echo can, or re-write it through a switch they have which has built in echo cans... Ugg.. Thanks, Billy - Original Message - From: Andrew Kohlsmith [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, July 01, 2004 10:56 AM Subject: Re: [Asterisk-Users] Echo cancellation, when software doesn't cut it. Whats next? On Thursday 01 July 2004 08:39, Robinson Tim-W10277 wrote: All voip providers will use digital 4-wire interconnect to Asterisk or similar, so echo problems are much reduced, as there are only 'echo points' at the far end and your handset. And on my PRI that is specifically where my echo is coming from... the far end. VOIP calls through nufone have no echo MOST PSTN calls through the PRI have no echo SOME PSTN calls (usually to local numbers NOT terminated at my local CO) have significant echo... I too have been unsuccessful in getting this zapped. My connection: Norstart MICS -- Adit600 --- T100P -- IAX2 -- TE405P -- Bell Canada PRI *1 = Xeon/2.4 with HT with T100P *2 = Xeon/2.4 with HT with TE405P *2 also does the NuFone IAX2 connection (it is always in the loop, as *1 is on a private network) Strange stuff, I am going to look at T1 echo cancellation hardware if I cant' get this solved. Tried: - echotraining=800 on *1 and *2 - echocancel=32,64,128 on both Eventually the MICS will have a digital connection to *1 instead of going through the Adit600 but we haven't got there yet :-) Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] Echo cancellation, when software doesn't cut it. Whats next?
I failed to mention I'm using a Cisco 2600 with Sip Re-invites.. and YES, I do have the echo can on the Cisco turned on, The echo is S bad, it's not even touching it... When we place 1-800 calls or call LD via our offnet provider, everything works fine, it's just with local calls on the PRI... We found out that the 1-800 #'s go out our carriers Sonus switch (a VoIP switch) which has 128ms Echo Can in it... Hmmm... Thanks, Billy - Original Message - From: Brent Franks [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, July 01, 2004 12:16 PM Subject: Re: [Asterisk-Users] Echo cancellation, when software doesn't cut it. Whats next? On Thu, 1 Jul 2004, Mike Benoit wrote: Obviously the less I spend the better. But if we have to, a few thousand more I guess. The problem I have is that this setup is more of a trial run. Once it works, I'm going to be cloning slightly smaller setups to 9 other cities. But they are pretty small, 1 or 2 lines and 2-4 phones in each location. I totally understand this. My users complain frequently about echo, and I am still unable to determine why sometimes it works great, other's it does not. The CPU and Memory are powerful enough to handle it, and we rarely ever see any load on the box. I too feel this is the major caveat to Asterisk right now. I am curious how anyone is achieving a near echo free system. We are shooting for 1 out of every 300 calls to have echo, which I think can be a realistic goal. Given the nature of open source, and the mix-and-match of components that come up, I can see where Digium is in a hard place to nail down the cause of every occurance. I will only be using POTS lines in each location. The current setup works great besides the echo, and some of the information I've read point to the Telco being the issue. If thats the case, I should in theory be able to get them to fix the problem. (though I could be dreaming) I think ultimately, if a Mediatrix box, or Cisco box can accomplish echo cancellation, Asterisk should be able to do it with as much success. Being that I am not an experienced Programmer, I try not to complain to loudly. With my level of involvement, I typically make the business case to customers and spec out ROI, etc. I do have a technical background, and am getting better at trouble shooting Asterisk and working on the source code. In fact, subscribing to the CVS list has taken me leap years ahead of understanding the changes and why they are being committed. I don't know how much more putting a DSP to handle echo can on the cards would cost, but if it were 400 - 500 more I would certainly pay it without a second thought, provided it worked. Echo, I think, is the largest draw back to VoIP, and will be the limit to entry into many businesses. I know my client, if they were to do it all over again, would choose a regular TDM (nortel, avaya) solution over the echo they are experiencing. I think asterisk is definitly headed in the right direction though, and nothing good comes over night. So everyone who has worked on it deserves to be commended. Without their insight and dedication, we wouldn't even be talking about this, or have alternatives to turn to. Regards, - Brent ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] anyone use mailboxexists?
Yes, I use it. Here's a sample extension of how to use it. exten = 1234,1,Answer() exten = 1234,2,MailboxExists(1234) exten = 1234,3,Dial(SIP/1234,20) ; Try to ring for 20 seconds, no answer goto voicemail exten = 1234,4,Voicemail(b1234) ; send to voicemail if busy exten = 1234,103,Dial(SIP/1234) ; Try to ring till answered exten = 1234,104,Busy() ; Give busy tone if busy. exten = 1234,204,Voicemail(u1234) ; send to voicemail if no answer - Original Message - From: Michael George [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, June 15, 2004 5:36 PM Subject: [Asterisk-Users] anyone use mailboxexists? I replied to a post of mine a few days ago asking of anyone uses mailboxexists(). I haven't received any replies. Perhaps few use it or perhaps the reply was overlooked. I thought I'd post the question one last time before giving up on it for now... Thanks! -Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] e164.org
'local' target? What's that?
- Original Message -
From: Matthew Asham [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, May 22, 2004 9:55 PM
Subject: Re: [Asterisk-Users] e164.org
You know, sleep deprivation cause people to do dumb things. The example
I pasted was hastily pasted and renumbered,
exten = _1NX,6,Playback(enum-lookup-failed)
exten = _1NX,7,Hangup
are actually:
exten = _1NX,103,Playback(enum-lookup-failed)
exten = _1NX,104,Hangup
Duane wrote up some more detailed examples at
http://www.e164.org/config.php.
Sorry for not proofing that when I posted it. I'll go sleep now.
On Sat, 2004-05-22 at 18:46, Tony Hoyle wrote:
Matthew Asham wrote:
; north america enum
exten = _1NX,1,Playback(doing-enum-lookup)
exten = _1NX,2,EnumLookup(${EXTEN})
exten = _1NX,3,BackGround(enum-lookup-successful)
exten = _1NX,4,Dial(${ENUM},30,tr)
exten = _1NX,5,Hangup
exten = _1NX,6,Playback(enum-lookup-failed)
exten = _1NX,7,Hangup
Interesting.. how does it know to go to '6', or does it just jump +4
on failure?
That reminds me I seriously need to restructure my extensions.conf...
there's
no way currently I could add anything like that without major surgery
(only
discovered the 'local' target this afternoon so I have everything
copied/pasted).
Tony
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Re: [Asterisk-Users] *** Asterisk sunday news: Read the sampleconfigs, Luke!
Mark, Would you please re-config or use a different mail client as to not send your replies back as attachments?? It's VERY kludgy, and, I'm just going to stop reading them.. along with all the other folks.. Thanks, Billy - Original Message - From: Mark Elkins [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, May 09, 2004 8:41 AM Subject: Re: [Asterisk-Users] *** Asterisk sunday news: Read the sampleconfigs, Luke! ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Cisco 7940 Phones as paging system?
That won't work.. That'll DIAL multiple phones/extensions, but will only
bridge 1 of them when it auto-answers..
What we need is a way to have something like meetme call multiple extensions
and bridge them to a meetme confrence (all of them muted but the admin of
course, as it's a one way page) and then we would have a true paging
system..
- Original Message -
From: Vic Cross [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, May 08, 2004 10:11 AM
Subject: Re: [Asterisk-Users] Cisco 7940 Phones as paging system?
On Fri, 7 May 2004, Ian A. Underwood wrote:
Joe Antkowiak wrote:
exten = 5101,1,Dial(SIP/5101,10,tA(intercom-tone))
exten = 5101,2,Congestion
That's not too bad, but how do you page a group of phones...like a real
intercom? That's what I'm dying to know!
in extensions.conf:
[globals]
INTERCOMLINES=SIP/Alice6SIP/Bob6SIP/Chuck6...
Then the extension is as per Joe's example, but replacing SIP/5101 with
${INTERCOMLINES}.
Extending this, you could set up various intercom numbers for different
parts of the office...
[globals]
SALESINTERCOM=SIP/Sales1-6SIP/Sales2-6...
MKTGINTERCOM=SIP/Marketing1-6...
...
[yourcontext]
exten = 5101,1,Dial(${SALESINTERCOM},10,tA(tone))
exten = 5102,1,Dial(${MKTGINTERCOM},10,tA(tone))
...
exten = 5110,1,Dial(${SALESINTERCOM}${MKTGINTERCOM}${...},10,tA(tone))
Cheers,
Vic Cross
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Re: [Asterisk-Users] 729 licence on scsi
SO, do you have a IDE CDROM? - Original Message - From: Mark Elkins [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, May 07, 2004 4:13 PM Subject: [Asterisk-Users] 729 licence on scsi ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Cisco 7940 Phones as paging system?
hey, can you send me the tone? - Original Message - From: Joe Antkowiak To: [EMAIL PROTECTED] Cc: [EMAIL PROTECTED] Sent: Friday, May 07, 2004 4:30 PM Subject: RE: [Asterisk-Users] Cisco 7940 Phones as paging system? This is what we have for this customer. They have five phones right now. Their normal extensions are 610x, but for intercom its 510x: exten = 5101,1,Dial(SIP/5101,10,tA(intercom-tone)) exten = 5101,2,Congestion If you want the wav file, let me know. If you make your own, be sure to put a 1-2 second pause in the beginning, because when the cisco answers it takes a second or to before it will send any audio to the speaker. -Original Message-From: mitchel [mailto:[EMAIL PROTECTED] Sent: Friday, May 07, 2004 4:16 PMTo: [EMAIL PROTECTED]Subject: RE: [Asterisk-Users] Cisco 7940 Phones as paging system? Hey Joe, Could I get a sample config for playing some intro tones on the intercom? I have the same thing but nobody is using it now because they are afraid of having someone call in and "listen in" so we need some way to announce the incoming intercom call. Thanks, MitchelJoe Antkowiak wrote: I am currently using 7960's with *, and line 6 is set to auto answer. Worksgreat, customer is happy. As far as an intro-tone, you can set the dialcommand to play a sound (using the announce option) before the call isconnected. I grabbed a simple tone wav file, and made it play that. Now,when the intercom ext is called, it plays the tone on the destination phone,and wa-la, intercomSo it works. Let me know if you need sample configs.-Original Message-From: [EMAIL PROTECTED][mailto:[EMAIL PROTECTED] On Behalf Of Philipp vonKlitzingSent: Friday, May 07, 2004 12:57 PMTo: [EMAIL PROTECTED]Subject: Re: [Asterisk-Users] Cisco 7940 Phones as paging system?Hi! able to support intercom/paging. Having searched the archives, it appears that this question was asked about 6 months ago, and the answer was that the Cisco phones support this using SCCP and having one line set to auto-answer, but at the time this was not supported in the SIP image. Is this still the case?Dunno about Cisco, but wanted to let you know that the recent Grandstream firmware (.55 and later) now also has an auto-answer option. Still I guess I should mention that the microphone of the GS phones in speakerphone mode is far from a brilliant implementation (- echo for the remote speaker talker, and too thin sound from the person in the room).Cheers, Philipp___Asterisk-Users mailing list[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users___Asterisk-Users mailing list[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users Do you Yahoo!?Win a $20,000 Career Makeover at Yahoo! HotJobs
Re: [Asterisk-Users] Fax Over VoIP
g711ulaw - Original Message - From: Michael Shuler [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, April 14, 2004 7:01 PM Subject: [Asterisk-Users] Fax Over VoIP Anyone know what protocols support a fax machine i.e. g.729, g.711, etc? Michael Shuler ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Bug with 'r' in dial
The lastest CVS's versions (both stable and head), the 'r' option in app_dial doesn't work with SIP and Re-invites. I've heard reports that it's not working with IAX2 either.. I'm using Cisco gateway and cisco ATA's and I am doing re-invites, and it's worked up till this point.. What's going on? Thanks, Billy +--+ | Billy Huddleston Senior Systems Administrator | | Net-Express http://www.nxs.net | | 114 Sherway Rd. Voice: 865-691-2011 | | Knoxville, TN 37922 Fax: 865-691-9894 | | [EMAIL PROTECTED]| +--+ ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Several H323 bugfixes - working SIP - H.323 translator
I just tried this, and it's not working for me.. I can't call a 2600 or a
CCM... What version of OpenH323 and PWLIB did you all use?
- Original Message -
From: Marian Durkovic [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, March 18, 2004 10:35 AM
Subject: [Asterisk-Users] Several H323 bugfixes - working SIP - H.323
translator
Hi all,
in an effort to create a SIP - H.323 translator we've found and fixed
several problems in H.323 channel. These inlcude:
for SIP-H.323 calls
- no ringback tone
- ringback not related to H.323 events
- one-way audio with Cisco CallManager
- incorrect Caller ID
for H.323-SIP calls
- not able to establish call with Cisco IOS 12.3(4)T
- ringback not related to SIP events
- no support for 183 Call Progress
- incorrect Caller ID
Please find the patches against aterisk 0.7.2 release below.
M.
--
Marian Durkovic network manager
Slovak Technical University Tel: +421 2 524 51 301
Computer Centre, Nam. Slobody 17 Fax: +421 2 524 94 351
812 43 Bratislava, Slovak RepublicE-mail/sip: [EMAIL PROTECTED]
--
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[Asterisk-Users] Codec negotation with re-invites..
I'm about over this.. okay,, here is what I got.. [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = inbound ; Default for incoming calls tos=lowdelay tos=184 disallow=all; Disallow all codecs allow=ulaw [gateway] type=friend host=1.1.6.9 canreinvite=yes qualify=yes dtmfmode=rfc2833 context=default disallow=all allow=ulaw allow=g729 [sipphoneg729] type=friend secret=password nat=yes host=dynamic canreinvite=yes qualify=200 context=longdistance-g729 dtmfmode=rfc2833 mailbox=2199 disallow=all allow=g729 [sipphoneulaw] type=friend secret=password nat=yes host=dynamic canreinvite=yes qualify=200 context=longdistance dtmfmode=rfc2833 mailbox=2199 disallow=all allow=ulaw okay, when I place a call from sipphoneulaw to the outside world via gateway, everything works fine.. If I place a call from sipphoneg729, it doesn't work.. One leg to the gateway will be ulaw, the leg to the phone will be g729, and, I have 1 way audio.. The sip phone can hear anything from the gateway, but, the gateway can't hear the phone. I've even went as far as to setup a seperate context for the g729 phone and do this.. ,SetVar,SIP_CODEC=g729 which, says it sets it to g729, but it's still a ulaw call.. Guys, this is a real problem... We're going be doing mixed configs.. and if a gateway says it can do both, and phone says it can only do one... then we should be using the compatable codec... PLEASE help.. This is going to cause problems in our rollout. Thanks, Billy +--+ | Billy Huddleston Senior Systems Administrator | | Net-Express http://www.nxs.net | | 114 Sherway Rd. Voice: 865-691-2011 | | Knoxville, TN 37922 Fax: 865-691-9894 | | [EMAIL PROTECTED]| +--+ ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Codec negotation with re-invites..
Okay, I add allow=g729 into the [general] section of sip.conf... I can now place calls via ulaw or g729 without any problems.. simply by setting the allow= in the phone's sip entry.. However, INBOUND is a whole nother problem... I get a really strange buzz sound on inbound calls.. and... here is a snippit of show sip channels while the call is in progress.. 1.1.1.24 8659342199 505b634c5cc 00103/0 0ms ms ULAW 1.1.1.29 8656914260 B392830B-17 00102/00102 0ms ms G729A .24 is the sip phone, .29 is the gateway I'm totally lost on this. - Original Message - From: Alex Volkov [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, March 12, 2004 3:23 PM Subject: Re: [Asterisk-Users] Codec negotation with re-invites.. Sounds to me that your asterisk first negotiates g729 with your phone, then negotiates ulaw with the gateway (since it *is* the preferred codec in your config), and on a re-invite the logic breaks up either in the phone or in the gateway (or perhaps in the asterisk itself, I am not absolutely clear on the details of re-invites). Try changing the order of codec preference for the gateway and see if that fixes your g729 phone and breaks the ulaw phone at the same time. Alex. - Original Message - From: Billy Huddleston [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, March 12, 2004 2:02 PM Subject: [Asterisk-Users] Codec negotation with re-invites.. I'm about over this.. okay,, here is what I got.. [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = inbound ; Default for incoming calls tos=lowdelay tos=184 disallow=all; Disallow all codecs allow=ulaw [gateway] type=friend host=1.1.6.9 canreinvite=yes qualify=yes dtmfmode=rfc2833 context=default disallow=all allow=ulaw allow=g729 [sipphoneg729] type=friend secret=password nat=yes host=dynamic canreinvite=yes qualify=200 context=longdistance-g729 dtmfmode=rfc2833 mailbox=2199 disallow=all allow=g729 [sipphoneulaw] type=friend secret=password nat=yes host=dynamic canreinvite=yes qualify=200 context=longdistance dtmfmode=rfc2833 mailbox=2199 disallow=all allow=ulaw okay, when I place a call from sipphoneulaw to the outside world via gateway, everything works fine.. If I place a call from sipphoneg729, it doesn't work.. One leg to the gateway will be ulaw, the leg to the phone will be g729, and, I have 1 way audio.. The sip phone can hear anything from the gateway, but, the gateway can't hear the phone. I've even went as far as to setup a seperate context for the g729 phone and do this.. ,SetVar,SIP_CODEC=g729 which, says it sets it to g729, but it's still a ulaw call.. Guys, this is a real problem... We're going be doing mixed configs.. and if a gateway says it can do both, and phone says it can only do one... then we should be using the compatable codec... PLEASE help.. This is going to cause problems in our rollout. Thanks, Billy +--+ | Billy Huddleston Senior Systems Administrator | | Net-Express http://www.nxs.net | | 114 Sherway Rd. Voice: 865-691-2011 | | Knoxville, TN 37922 Fax: 865-691-9894 | | [EMAIL PROTECTED]| +--+ ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Stange notices and Warnings..
I've been getting a few strange things latly.. Mar 1 14:44:07 NOTICE[1142106560]: chan_sip.c:5585 handle_request: Registration from 'sip:[EMAIL PROTECTED];user=phone' failed for '63.169.60.253' -- Registered SIP '2767069017' at 63.169.60.253 port 5060 expires 120 This is a Cisco ATA running 3.1, I've got several others, but this one is the only one that does it... Thanks, Billy +--+ | Billy Huddleston Senior Systems Administrator | | Net-Express http://www.nxs.net | | 114 Sherway Rd. Voice: 865-691-2011 | | Knoxville, TN 37922 Fax: 865-691-9894 | | [EMAIL PROTECTED]| +--+ ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Problems with ATA's locking up..
That's just it, I'm not doing anything.. Just normal use.. as far as I can tell, they end up locking up with or without anyone using them as far as I can tell.. Thanks, Billy - Original Message - From: Florian Overkamp [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, February 08, 2004 12:08 PM Subject: Re: [Asterisk-Users] Problems with ATA's locking up.. Hi, Citeren Billy Huddleston [EMAIL PROTECTED]: Using them with * and a Cisco GW, they're on PUBLIC ip's, No Firewall or anything, I am using re-invites. Pretty standard setup. When they lockup, you can't ping them, or get to the http interface, and I even think the IVR stops responding when you push the button. Yes, but is anything specific happening when they hang ? (What are you doing that seems to cause the hang ?) I have pretty similar setups, so I could try to recreate your scenario ? Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] Problems with ATA's locking up..
http://www.nxs.net/cisco_ata_186.htm - Original Message - From: CW_ASN [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, February 08, 2004 12:40 PM Subject: Re: [Asterisk-Users] Problems with ATA's locking up.. Could you share your 3.0.0 config? - Original Message - From: Florian Overkamp [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, February 08, 2004 2:10 PM Subject: Re: [Asterisk-Users] Problems with ATA's locking up.. Hi, Citeren CW_ASN [EMAIL PROTECTED]: 3.0.0 have some problems. Sometimes, ata answers to invite with Not found or Busy here. This is a strange behavior. I'm using now 2.16.2 Hm ? I have not seen this happening yet. 2.16 has alternative behaviour regarding flash transfers... Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problems with ATA's locking up..
Anyone had any problems with ATA's running 3.0 software locking up? Thanks, Billy +--+ | Billy HuddlestonSenior System Administrator | | Net-Express http://www.nxs.net | | 114 Sherway Rd. Voice: 865-691-2011 | | Knoxville, TN 37922 Fax: 865-691-9894 | | [EMAIL PROTECTED]| +--+ ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] voip phones
and with the HT-286 you get a Chinaman in a box! :) - Original Message - From: Michael Koehler [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, February 03, 2004 5:45 PM Subject: Re: [Asterisk-Users] voip phones I prefer SIP Phones, Grandstream BT-100 IP-Phone or the HT-286 Analogue telephone adapter Why? - brilliant user interface, with or with out a web browser - cristal clear voice even with low band codecs - PPP over ethernet (PPPoE) aware - continual firmware improvement - plenty of tweak options - economically priced - protocol conform - made in china - fast shipping Retail from $39 to $245 .. google is your friend. Tim Sailer wrote: What is the best inexpensive voip phone out there? I want to try a few with *, but don't want to go broke while I'm just playing around... Tim ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] canreinvite and codec negotations...
Okay, now on to my problem.. I have people who will be using ulaw, and I have people who will be using g729.. I want to set it up so that canreinivte will work.. I have a single cisco gateway.. Asterisks isn't handling the negotation between the 2 devices very well.. For example.. [gateway] type=friend host=1.2.3.4 canreinvite=yes qualify=200 dtmfmode=rfc2833 context=default disallow=all allow=ulaw allow=g729 [123] type=friend secret=abc nat=yes host=dynamic canreinvite=yes qualify=200 context=default dtmfmode=rfc2833 mailbox=2199 callerid=Joe Blow 123-456-7890 disallow=all allow=g729 [321] type=friend secret=abc nat=yes host=dynamic canreinvite=yes qualify=200 context=default dtmfmode=rfc2833 mailbox=321 callerid=Joe Blow 321-456-7890 disallow=all allow=ulaw Okay, in this configs, gateway would be my cisco 26xx gateway.. ext 123 would be a g729 customer.. and 321 would be a ulaw customer. When someone calls ext 123, the cisco send the call to asterisk, it *WILL* use ulaw... then it will initiate a call to g729, well... Now we have a codec mismatch, and canreinvite won't work... EVEN though gateway can do g729.. ext 321 won't have any problems.. It'll work fine for them.. What can we do to get this to work like it should? Thank, Billy +--+ | Billy HuddlestonSenior System Administrator | | Net-Express http://www.nxs.net | | 114 Sherway Rd. Voice: 865-691-2011 | | Knoxville, TN 37922 Fax: 865-691-9894 | | [EMAIL PROTECTED]| +--+ ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] G.729 Licenses from Digium
IDE/SCSI interfaces, SCSI only installed, WITH IDE CDROM installed with CDROM in drive. - g729 WILL WORK. I'm running a system right now with 24 licences.. Tested it with a single license before purchasing the other 23. You MUST have a CDROM in the drive when you run the install program.. and MUST have it in the drive when you bring Asterisk's up. Thanks, Billy - Original Message - From: Senad Jordanovic [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, January 21, 2004 5:28 PM Subject: RE: [Asterisk-Users] G.729 Licenses from Digium zoa wrote: This is absolutely not true. I have 3 (raid) scsi asterisk machines in production. Joachim. At 11:32 21/01/2004 -0500, you wrote: In my view at least one IDE drive must be installed in order for * g729 license to work. To simplyfy, here is the matrix (This is how I think it is please confirm) IDE Disk Install - g729 coder work. IDE/SCCI interfaces. Only a SCSI disk installed - g729 will not work. IDE/SCSI Interfaces. At lease one IDE disk installed - g729 will work. SATA Serial ATA Disk I have no clue how it works. Is SATA considered a IDE disk or a SCSI disk ? This is an issue that VoiceAge need to address soon. - SamW -Original Message- From: Amaury Jacquot [mailto:[EMAIL PROTECTED] Sent: Wednesday, January 21, 2004 4:32 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] G.729 Licenses from Digium Terence Parker wrote: OK - but what counts as a SCSI system? These days there are lots of pseudo-SCSI systems around - such as our server which runs a serial-ATA RAID but the driver is loaded as a SCSI device. Is that still IDE? Or SCSI? technically, it uses the SCSI command set over a serial link, so, it's SCSI Terence I know one thing for sure... G729 WILL NOT WORK after installation *(it never realy installs but does the segmentation faults), * will not start, and you will need to prevent g729 module from Starting in order for * to start. So do not buy if your box is SCSI in any part. Ta SJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Can you please clarify which part are you referring as not being true? ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] Handytone 286 - calling out
I've not had ANY problems using info OR rfc2833.. I did have problems using inband. Try switching to it and see how it works.. I NEVER had a problem with double digits, and, I believe that the reference to GS phones having that problem with * was retracted. Thanks, Billy - Original Message - From: Senad Jordanovic [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, November 26, 2003 4:14 AM Subject: RE: [Asterisk-Users] Handytone 286 - calling out Billy Huddleston wrote: change dtmf to info on both * and in the handytone. - Original Message - From: Senad Jordanovic [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, November 25, 2003 8:01 PM Subject: [Asterisk-Users] Handytone 286 - calling out Hi, Just received recently released Grandstream handytone 286 ATA for testing. I can call ATA from any other extensions and conversations seems to be of quite good quality. However placing calls from ATA is not possible at all to any extensions. After dialing there no indications of any kind from ATA at all. It just hangs in there. ATA is behind NAT, registers to an * with public IP with no problems and it uses 1.0.4.17 firmware. Web config screen has detected firewall/NAT type is open Internet as network firewall. Here is my sip.conf: [2202] callerid=HandyTone 2202 username=2202 context=intern qualify=500 type=friend secret=XX host=dynamic dtmfmode=inband canreinvite=no reinvite=no disallow=all allow=ulaw allow=alaw Any suggestions/pointers will be appreciated. Ta SJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users My understanding from this months GS related posts is that info is not sending the digits properly. Is that the case with you? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Handytone 286 - calling out
change dtmf to info on both * and in the handytone. - Original Message - From: Senad Jordanovic [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, November 25, 2003 8:01 PM Subject: [Asterisk-Users] Handytone 286 - calling out Hi, Just received recently released Grandstream handytone 286 ATA for testing. I can call ATA from any other extensions and conversations seems to be of quite good quality. However placing calls from ATA is not possible at all to any extensions. After dialing there no indications of any kind from ATA at all. It just hangs in there. ATA is behind NAT, registers to an * with public IP with no problems and it uses 1.0.4.17 firmware. Web config screen has detected firewall/NAT type is open Internet as network firewall. Here is my sip.conf: [2202] callerid=HandyTone 2202 username=2202 context=intern qualify=500 type=friend secret=XX host=dynamic dtmfmode=inband canreinvite=no reinvite=no disallow=all allow=ulaw allow=alaw Any suggestions/pointers will be appreciated. Ta SJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] strange SIP authentication/authorization behaviour
loose username=ipphone9 Not needed.. the [109] is really the username - Original Message - From: Anton Yurchenko [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, November 24, 2003 11:42 AM Subject: [Asterisk-Users] strange SIP authentication/authorization behaviour When I have an ip hardphone username setup in my sip.conf : [109] type=friend username=ipphone9 secret=bla-la host=dynamic dtmfmode=rfc2833; Choices are inband, rfc2833, or info defaultip=172.20.0.139 mailbox=109 ; Mailbox for message waiting indicator callerid=ipphone9 109 callgroup=1 pickupgroup=1 and this user has a wrong password then calls are denied, but when I just change the userID on the phone to a nonexistant for example 110, the calls go through ! though I see on the console messages about wrong authentication. I`m running a CVS version from friday. Thanks, -- Anton Yurchenko[EMAIL PROTECTED] Digital Generation ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] tunnel iax via gnophone with ssh?
Use CIPE, It's a UDP based VPN solution. - Original Message - From: Alastair Maw [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, November 20, 2003 1:37 PM Subject: Re: [Asterisk-Users] tunnel iax via gnophone with ssh? On 20/11/03 15:44, Chris Hirsch wrote: Hey all...I'm trying to use gnophone to connect to my asterisk box behind my firewall..I thought I could just setup a tunnel with something like ssh host.com -L5036:asteriskserver:5036 and just change my gnophone to connect to localhost:5036 but I never see anything happen on the asterisk server. I'm even trying this on the same network just in case there is something funky with NAT. Anybody have any ideas? Yes, IIRC SSH only tunnels TCP. IAX is UDP based. You'll need to find something that will tunnel UDP over TCP, so you can tunnel that over SSH (!). Good luck. :) Alastair ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] snatching calls
how could you do this with sip and VOIP? - Original Message - From: Steven Critchfield [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, November 04, 2003 5:40 PM Subject: Re: [Asterisk-Users] snatching calls On Tue, 2003-11-04 at 15:28, Shoval Tomer wrote: Hi, Our current PBX (Panasonic) has a setting that enables users to snatch calls ringing at other extensions. I'm not sure snatching is a correct term for this so let me elaborate. Let's say you sit in a room with five other people. Each one has it's own extension. One person goes out. As soon as he leaves the room his phone starts ringing. The other guys in the room want to answer his phone for him and take a message, but they won't get up for it. I just pick up our extension, hit *40, and the call is automatically transferred to my extension. Is this doable with Asterisk? Is it possible to divide extensions into groups like a group per room so I won't snatch a call from another room in the building, that I wasn't even aware was ringing You want to look into call groups and pickup groups. To pickup the call you use *8#. from /usr/src/asterisk/configs/zapata.conf.sample ; ; Ring groups (a.k.a. call groups) and pickup groups. If a phone is ringing ; and it is a member of a group which is one of your pickup groups, then ; you can answer it by picking up and dialing *8#. For simple offices, just ; make these both the same ; callgroup=1 pickupgroup=1 -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
