Re: [asterisk-users] iaxmodem, chan_capi, hylafax problem and faxing in general
Hello Faris, Only I've sidetracked and am currently trying to use capi4hylafax instead of iaxmodem which seems to work wonderfully but I'm having some issues with root verses uucp permissions which is spoiling my fun. Make sure not to run faxgetty together with capi4hylafax. -- Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Hook flash time problem on TDM400/FXS
I have been trying for some time now to make the hook flash work on the FXS port. I am using Asterisk 1.4.10.1 with zaptel 1.4.4. When I manually flash the hook I can manage to find the duration to put a call on hold. However when pushing the flash button it never works. The phone's flashtime seems to be too short. I tried to set a shorter flashtime in the zapata.conf file, but it seems to be ignored. I have flash=100 configured in the zapata.conf en when reloading it, this is what is reported on the asterisk console: == Parsing '/etc/asterisk/zapata.conf': == Parsing '/etc/asterisk/mgcp.conf': Found Found [Aug 17 08:27:01] WARNING[15818]: chan_zap.c:7 process_zap: Ignoring flash [Aug 17 08:27:01] WARNING[15818]: chan_zap.c:7 process_zap: Ignoring signalling -- Reconfigured channel 1, FXS Kewlstart signalling [Aug 17 08:27:01] WARNING[15818]: chan_zap.c:7 process_zap: Ignoring signalling -- Reconfigured channel 2, FXO Kewlstart signalling [Aug 17 08:27:01] WARNING[15818]: chan_zap.c:7 process_zap: Ignoring signalling -- Reconfigured channel 3, FXO Kewlstart signalling [Aug 17 08:27:01] WARNING[15818]: chan_zap.c:7 process_zap: Ignoring signalling -- Reconfigured channel 4, FXO Kewlstart signalling How can I adjust the flashtime on a FXS port in such a way that I can use the flash function of the phone connected to it. This is especially important for a DECT phone for which I cannot do a manual hook flash. Without this I cannot transfer calls. My searches on the internet did not give me any information other than that I should change the flash time parameter in zapata.conf. An old message indicated that I should change some .h file and recompile zaptel drivers, but could not find the particular piece of code probably because it has changed considerably later on. Also the information I found indicated that the flash time in Europe is generally very short (80 - 120 ms) as compared to the US (750ms ?). I am in the Netherlands. TIA, Hans Feringa ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem in installing libmfcr2 for configuring MFC/R2
Hi, I am trying to configure for MFC/R2 for asterisk. With the help of one of the asterisk users group member patrick I am able to install libunicall library. Now, when trying to install libmfr2-0.0.3 it is giving error. On running running command $./configure It is giving error - can't build without libtiff I downloaded tiff-3.8.2 and installed the libraries but still the problem persists. Can anybody will tell me how to remove the problem. Thanx and regards sanchal ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RAW asterisk!
On Thu, 16 Aug 2007, Bill Andersen wrote:
OK, I understand that. But if I gotta learn how to support
myself to do advanced features, why pay them at all? I'll
just become my own expert :()
That's how I started...
Sit-down and work out what features you want - and do you want them
supported in the phones or in the system - eg. features like divert or do
not disturb - many phones have this feature, but they need to be turned on
for them to work, or you can implement them in the PBX itself. Similarly
for speed dialling, etc.
I took the approach that if I could make the PBX work with the dumbest of
phones then it'd work for any phone regardless of it's features, so
implemented a whole raft of star codes in the PBX itself.
So to implement do not disturb, I have:
; *490: *491:
; Clear/Set Do Not Disturb
exten = *490,1,Answer()
exten = *490,n,Macro(clearStarCode,doNotDisturb)
exten = *491,1,Answer()
exten = *491,n,Set(dndCode=${EXTEN:3})
exten = *491,n,Set(DB(${CALLERID(num)}/doNotDisturb)=${dndCode})
exten = *491,n,Macro(starAck)
but then you need to handle it in the bit that handles calls internally, so
mine looks like:
; Check for Do Not Disturb
exten = s,n,Set(DND=${DB(${MACRO_EXTEN}/doNotDisturb)})
exten = s,n,GotoIf(${DND}?:doneDoNotDisturb)
exten = s,n,Wait(90)
exten = s,n,Hangup()
exten = s,n(doneDoNotDisturb),Noop(Carrying on after DO NOT DISTURB Check)
and so on ...
There are other ways to do this, and other better/different? macros to
handle dialling and so on, or you might want to do it all in AGI, AEL,
or...
My systems are completely done in dialplan as I've found it adequate for
my needs and this includes setting up IAX trunks between offices which
work out the correct site caller id without using DUNDi
I just googled searched through the books and voip-wiki for dialplan
examples and built it up from there.
Keep It Simple is my motto and I'm not putting a MySQL database on a
diskless system when it's not needed... Over complexity for the sake of
nerdyness really peeves me! (But it could be argued that that's because I
was brought up on trying to squeeze programs into 256 byte of RAM too many
years ago ;-)
Enjoy!
Gordon
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Re: [asterisk-users] Problem in installing libmfcr2 for configuring MFC/R2
On Fri, Aug 17, 2007 at 01:23:11PM +0530, [EMAIL PROTECTED] wrote: Hi, I am trying to configure for MFC/R2 for asterisk. With the help of one of the asterisk users group member patrick I am able to install libunicall library. Now, when trying to install libmfr2-0.0.3 it is giving error. On running running command $./configure It is giving error - can't build without libtiff I downloaded tiff-3.8.2 and installed the libraries but still the problem persists. Can anybody will tell me how to remove the problem. Thanx and regards sanchal Look in config.log (near the end). What command fails that causes this error? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 1.4.10.[0,1] crashes when call parked
100% repeatable (for me). Sip phone A calls Sip phone B. Either Sip phone A or B does #700. The party that keyed #700 gets the parked announcement (eg 701) and the other party get MOH. There is still an audio channel between the two SIP phones at this point. When the party that typed #700 hangs up, Asterisk crashes. This has been working in previous 1.4's (but not 1.4.10) and I havn't changed my parking config. Here's what comes up on the console as it crashes. -- SIP/Testsnom-00709570 Playing 'digits/7' (language 'en') -- SIP/Testsnom-00709570 Playing 'digits/0' (language 'en') -- SIP/Testsnom-00709570 Playing 'digits/1' (language 'en') -- Added extension '701' priority 1 to parkedcalls -- Stopped music on hold on SIP/115-0072f7a0 == SIP/115-0072f7a0 got tired of being parked == Spawn extension (macro-stdsip, s, 6) exited non-zero on 'SIP/Testsnom-00709570' in macro 'stdsip' == Spawn extension (macro-stdsip, s, 6) exited non-zero on 'SIP/Testsnom-00709570' *** glibc detected *** asterisk: double free or corruption (out): 0x2ab23a7ed9f0 *** === Backtrace: = /lib/libc.so.6[0x2ab23a620733] /lib/libc.so.6(__libc_free+0x84)[0x2ab23a6208b4] asterisk(ast_channel_free+0xf6)[0x438fa6] asterisk(ast_hangup+0x35a)[0x43b84a] /usr/lib/asterisk/modules/res_features.so[0x2b8298c0] asterisk[0x4a719c] /lib/libpthread.so.0[0x2ab239da23ca] /lib/libc.so.6(__clone+0x6d)[0x2ab23a67f55d] === Memory map: Anyone got any ideas? -- Regards, Russell | Russell Brown | MAIL: [EMAIL PROTECTED] PHONE: 01780 471800 | | Lady Lodge Systems | WWW Work: http://www.lls.com | | Peterborough, England | WWW Play: http://www.ruffle.me.uk | ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Lock extension from asterisk
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi all I am working in a new set up with Grandstream GXP-2000 handsets. I like those phone, but they lack a feature I need: the phone cannot be locked by the user. What I actually want is a user to be able to avoid someone else making calls from his phone without giving him access to SIP configuration access to the phone. i.e. let say I want user 132 (that uses extension 132 in our system) to be able to lock his phone (located in a publicly accessible office). Could he dial an special extension (i.e. ) and Asterisk will drop any call until another special extension (i.e. ) is dialed? Suggestions? - -- Andres Jimenez -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.3 (GNU/Linux) Comment: http://firegpg.tuxfamily.org iD8DBQFGxX6A8SZxpGYWwpYRAtojAJ4yKE77nv9rpkoXXr1i4SOiLPb7JACgug+7 64yg8fCDRdnmeZFmmpGynwQ= =eCWP -END PGP SIGNATURE- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dialplan / AGI autoanswer question
Haven't tried with new 1.4 branch of Asterisk/Zaptel, but in 1.2, if correctly configured, there was no need to Wait(x) to let zaptel to get the CID on analog lines: it was zaptel itself to not let the call go through the dialplan until the second ring. I think it shoud be something like: usecallerid=yes callerid=asreceived in zapata.conf for relevant channels. Hope this helps... Regards picciux 2007/8/16, Matthew Harrell [EMAIL PROTECTED]: Thanks. I was hoping there might be a way to detect whether the CID routine was done or not. I've still seen occasions where it wasn't available for callers that I know had it. Maybe my phone service is just a little slow sometimes Wait(2) is what I do. Matthew Harrell wrote: First, it seems I have to have a 2 - 3 second wait before the AGI call in order to get valid CID data. Usually 2 seconds suffices for this one setup but during that time the caller has had two rings before the local extension has even begun to ring. Is there something I am doing wrong that causes it to take so long to get the CID? CallerID info is sent between the first and second ring. Well that would explain that problem, wouldn't it? Is there a proper way to wait for the CID data to be filled in if available or is Wait(2) my best option? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Matthew Harrell Never underestimate the power of Bit Twiddlers, Inc. very stupid people in large groups. [EMAIL PROTECTED] ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk multiport
Steven wrote: I am curious. Why would one need to do this? If a phone connect to 5060 from another port number, asterisk happily works, so why use multiple port on asterisk? I cannot see the thread history but from the context, I would say because many ISPs block 5060, 25, and others. Thanks, Steve Totaro ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk multiport
What i actually do is make asterisk listen on some other port like 5097 and redirect port 5060 to it with iptables like this /sbin/iptables -t nat -A PREROUTING -i eth0 -p udp --dport 5060 -j DNAT --to YOURIPHERE:5097 This works very well . If i make asterisk listen on 5060 and redirect say 5097 to 5060 i had lot of problems with firewalled systems ( blocked 5060 by isp ) . Also on blocked end its recommended to use some softphone like xlite which completely allows you to set custom ports on machine itself to listen, taking 5060 completely out of picture . On 17/08/07, Steve Totaro [EMAIL PROTECTED] wrote: Steven wrote: I am curious. Why would one need to do this? If a phone connect to 5060 from another port number, asterisk happily works, so why use multiple port on asterisk? I cannot see the thread history but from the context, I would say because many ISPs block 5060, 25, and others. Thanks, Steve Totaro ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] A102 card, BT ISDN30e, silence
Summary:
Can't hear incoming/outgoing calls to/from ISDN over Sangoma A102 card.
Debian stable + Asterisk 1:1.2.13.
Thanks for the response, Andres. We've changed the timing source but
still no joy. This is very odd; calling between internal AIX/SIP
extensions works fine.
If anyone else can think of anything, I would be grateful to hear.
Rory
On 16/08/07, Andres Paglayan ([EMAIL PROTECTED]) wrote:
On Aug 16, 2007, at 8:58 AM, Rory Campbell-Lange wrote:
However both incoming and outgoing calls are greeted by silence.
...
Zaptel
--
loadzone=uk
defaultzone=uk
#Sangoma A101 port 1 [slot:3 bus:2 span: 1]
span=1,0,0,ccs,hdb3,crc4
the first 0 is the timing source,
if you are connecting this span to a telco
it should be 1 for your telco to provide timing,
which is what they expect
; note that only 25 channels are active
bchan=1-15,17-31
dchan=16
Zapata
--
-
[channels]
language=en
context=from-pstn
signalling=pri_cpe
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=no
echotraining=800
rxgain=0.0
txgain=0.0
group=0
callgroup=1
pickupgroup=1
immediate=no
pridialplan=unknown
prilocaldialplan=unknown
group=1
callerid=asreceived
;Sangoma A101 port 1 [slot:3 bus:2 span: 1]
switchtype=euroisdn
context=from-pstn
group=0
signalling=pri_cpe
channel = 1-15,17-31
Extensions
--
-
[general]
static=yes
writeprotect=no
autofallthrough=yes
clearglobalvars=no
priorityjumping=no
[globals]
CONSOLE=Console/dsp ; Console interface for demo
IAXINFO=guest ; IAXtel username/password
TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0)
[dundi-e164-local]
switch = DUNDi/e164
[local]
ignorepat = 9
include = default
include = parkedcalls
[default]
exten = 001,1,Dial(SIP/001,5)
exten = 001,n,VoiceMail([EMAIL PROTECTED])
exten = 001,n,Playback(vm-goodbye)
exten = 001,n,Hangup
exten = 901,1,Dial(IAX2/901,5)
exten = 901,n,Hangup
exten = 100,1,Dial(SIP/100,5)
exten = 100,n,VoiceMail([EMAIL PROTECTED])
exten = 100,n,Playback(vm-goodbye)
exten = 100,n,Hangup
exten = _0XX,1,Dial(Zap/g0/${EXTEN})
exten = _ZX,1,Dial(Zap/g0/${EXTEN})
[from-pstn]
; BT pass in 6 digits only?
exten = 032685,1,Answer()
exten = 032685,2,Playback(demo-echotest)
exten = 032685,3,Echo()
exten = 032685,4,Playback(demo-echodone)
exten = 032685,5,Hangup()
--
Rory Campbell-Lange
Director
Campbell-Lange Workshop Ltd.
[EMAIL PROTECTED]
www.campbell-lange.net
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Andres Paglayan
--Harmony is more important than being right
Bapak
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--
Rory Campbell-Lange
Director
Campbell-Lange Workshop Ltd.
[EMAIL PROTECTED]
www.campbell-lange.net
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Re: [asterisk-users] 99 bottles of beer
On Thu, 2007-08-16 at 19:38 -0600, Steve Murphy wrote:
On Thu, 2007-08-16 at 07:56 -0400, Russell Bryant wrote:
Gordon Henderson wrote:
; *99:
; 99 bottles of beer on the wall.
exten = *99,1,Noop(99 Bottles of beer on the wall)
exten = *99,n,Answer()
exten = *99,n,Set(bottles=99)
exten = *99,n(loop),Noop(There are ${bottles} bottles of beer on the
wall)
exten = *99,n,SayNumber(${bottles})
exten = *99,n,Noop(Take one done and pass it round and there's)
exten = *99,n,Set(bottles=$[${bottles}-1])
exten = *99,n,Noop(${bottles} bottles of beer on the wall)
exten = *99,n,SayNumber(${bottles})
exten = *99,n,GotoIf($[${bottles} 0]?loop)
exten = *99,n,Noop(We're out of beer!)
exten = *99,n,Hangup()
Too much dial plan mashing this morning and I rememberd this site:
http://99-bottles-of-beer.net/
And now, in AEL! (This is untested, I just wanted to see how it would
look.)
context silly {
*99 = {
NoOp(99 Bottles of beer on the wall);
Answer();
bottles=99;
while (${bottles} 0) {
NoOp(${bottles} bottles of beer on the wall, ${bottles} bottles of
beer);
SayNumber(${bottles});
NoOp(Take one down, pass it around);
bottles=${bottles} - 1;
NoOp(${bottles} bottles of beer on the wall);
}
NoOp(We're out of beer!);
Hangup();
}
}
Lol, Well done, Russell!
How about this one: from an extensions.conf that someone posted on the
internet, I think, and I converted to AEL; I'm sorry, but I can't find
the original author.
(If anybody can find his post, I'd love to give him credit.) I did test
this out,
and it works; just put a call to the macro ( guessgame(); ) in an
extension in your dialplan
macro guessgame()
{
startpoint:
while (1)
{
Playback(guessit/intro);
set(GUESS=);
GUESS=${EPOCH}%9;
Set(TIMEOUT(digit)=3);
Set(TIMEOUT(response)=5);
while (1)
{
Read(NUMBER,guessit/input_number,1);
Verbose(Got ${NUMBER} from Read);
if( ${NUMBER} = * || ${NUMBER} = # ||
${NUMBER} = )
{
Playback(guessit/thatsnotanumber);
}
else if (${NUMBER} = ${GUESS})
{
Playback(guessit/win);
break; // the only way out of this loop!
}
else if (${NUMBER} ${GUESS})
{
Playback(guessit/less);
}
else if (${NUMBER} ${GUESS})
{
Playback(guessit/more);
}
else /* what other stuff can the user enter than a
number, #,
nothing, or * ? */
{
Playback(guessit/thatsnotanumber);
}
}
/* You get here after a successful guess */
Wait(.5);
Read(AGAIN,guessit/playagain,1);
if (${AGAIN} != 1)
break;
}
Playback(guessit/goodbye);
return;
catch t
{
playback(guessit/goodbye);
return;
}
catch i
{
playblack(invalid);
}
}
murf
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Hey murf,
here is the link for the credit,
http://www.voipphreak.ca/archives/358-Asterisk-Howto-Numbers-Guessing-Game.html
its also in the wiki examples.
http://www.voip-info.org/wiki/view/AEL+Example+Snippets
db
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Re: [asterisk-users] Lock extension from asterisk
On Fri, 17 Aug 2007, Andres Jimenez wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi all
I am working in a new set up with Grandstream GXP-2000 handsets. I
like those phone, but they lack a feature I need: the phone cannot be
locked by the user.
What I actually want is a user to be able to avoid someone else making
calls from his phone without giving him access to SIP configuration
access to the phone.
i.e. let say I want user 132 (that uses extension 132 in our system)
to be able to lock his phone (located in a publicly accessible
office).
The easy answer is: Yes...
Could he dial an special extension (i.e. ) and Asterisk will drop
any call until another special extension (i.e. ) is dialed?
Suggestions?
It all depends on how your dialplan works. If you have one macro that
controls calls from extensions to other extensions, or outside lines, then
you can implement 2 numbers to set/clear a flag in the astdb, then in
the bits where to call other extensions, (or outside lines) call a macro
that tests for the flag being set...
You can use the extensions voicemail PIN to validate the unlocking too for
futher security.
S (all untested!)
exten = ,1,Answer()
exten = ,n,Set(me=${CALLERID(num)})
exten = ,n,Set(DB(${me}/locked)=1)
exten = ,1,Answer()
exten = ,n,Set(me=${CALLERID(num)})
exten = ,n,VMAuthenticate(${me})
exten = ,n,Set(DB(${me}/locked)=)
(I think I swapped the and here, but I'm sure you can see that!)
and in the dial-plan where call processing takes place:
exten = s,1,Set(me=${CALLERID(num)})
exten = s,n,Set(locked=${DB(${me}/locked)})
exten = s,n,GotoIf(${locked}?:doneLockCheck)
exten = s,n,Playback(sorry-cant-let-you-do-that)
exten = s,n,Hangup()
exten = s,n(doneLockCheck),Noop(We're not locked)
Gordon
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Re: [asterisk-users] asterisk multiport
Ahh, I see. Good point. -- -- Steven http://www.glimasoutheast.org Steve Totaro [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] Steven wrote: I am curious. Why would one need to do this? If a phone connect to 5060 from another port number, asterisk happily works, so why use multiple port on asterisk? I cannot see the thread history but from the context, I would say because many ISPs block 5060, 25, and others. Thanks, Steve Totaro ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] No audio on ISDN PRI calls
Hello, I have a Sangoma A101 connected to an ISDN30 (E1 in the UK) with some Snom 300 and Idefisk softphones. I can do SIP and IAX2 calls just fine, however I cant get any audio in either direction on the Zap channels. When I call in or dial out over the ISDN30 (UK E1) I can see the call answered/placed on the CLI and then silence follows. I've been provisioned 25 out of the 31 channels only at the moment and I'm using a dummy number given by BT for testing. The is no alarms on the line, nor errors on the CLI or in the /va/log/messages or wanrouter. Any ideas? Thank you much. Veselin -- Regards, Veselin Kantsev [EMAIL PROTECTED] Campbell-Lange Workshop ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.4.10.[0,1] crashes when call parked
Russell Brown wrote: 100% repeatable (for me). Sip phone A calls Sip phone B. Either Sip phone A or B does #700. The party that keyed #700 gets the parked announcement (eg 701) and the other party get MOH. There is still an audio channel between the two SIP phones at this point. When the party that typed #700 hangs up, Asterisk crashes. This has been working in previous 1.4's (but not 1.4.10) and I havn't changed my parking config. Here's what comes up on the console as it crashes. -- SIP/Testsnom-00709570 Playing 'digits/7' (language 'en') -- SIP/Testsnom-00709570 Playing 'digits/0' (language 'en') -- SIP/Testsnom-00709570 Playing 'digits/1' (language 'en') -- Added extension '701' priority 1 to parkedcalls -- Stopped music on hold on SIP/115-0072f7a0 == SIP/115-0072f7a0 got tired of being parked == Spawn extension (macro-stdsip, s, 6) exited non-zero on 'SIP/Testsnom-00709570' in macro 'stdsip' == Spawn extension (macro-stdsip, s, 6) exited non-zero on 'SIP/Testsnom-00709570' *** glibc detected *** asterisk: double free or corruption (out): 0x2ab23a7ed9f0 *** === Backtrace: = /lib/libc.so.6[0x2ab23a620733] /lib/libc.so.6(__libc_free+0x84)[0x2ab23a6208b4] asterisk(ast_channel_free+0xf6)[0x438fa6] asterisk(ast_hangup+0x35a)[0x43b84a] /usr/lib/asterisk/modules/res_features.so[0x2b8298c0] asterisk[0x4a719c] /lib/libpthread.so.0[0x2ab239da23ca] /lib/libc.so.6(__clone+0x6d)[0x2ab23a67f55d] === Memory map: Anyone got any ideas? No ideas but I also found something last night that could be related. I have the following: Home SIP -SIP- Home Asterisk -IAX- Work Asterisk Calls from Home SIP to Home Asterisk (like vms) sound fine. Calls from Home SIP through Home Asterisk to Work Asterisk sound horrible on the SIP side. I don't know how it sounded on the Work side since I was checking voicemail. It occurred on every call all the time with 1.4.10 and 1.4.10.1. Reverting Home Asterisk to 1.4.9 solved the problem. Maybe something got fixed in chan_sip in 1.4.10? -Dave ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Lock extension from asterisk
Gordon Henderson wrote:
On Fri, 17 Aug 2007, Andres Jimenez wrote:
exten = ,1,Answer()
exten = ,n,Set(me=${CALLERID(num)})
exten = ,n,Set(DB(${me}/locked)=1)
exten = ,1,Answer()
exten = ,n,Set(me=${CALLERID(num)})
exten = ,n,VMAuthenticate(${me})
exten = ,n,Set(DB(${me}/locked)=)
This is good, but it can be done with just 1 extension:
exten = 80*,1,Answer()
exten = 80*,2,Set(LOCKED=${DB(phonelocked/${CALLERID(number)})})
exten = 80*,3,GotoIf($[${LOCKED} = YES]?80*,4:80*,101)
exten = 80*,4,Set(DB(phonelocked/${CALLERID(number)})=NO)
exten = 80*,5,Playback(de-activated)
exten = 80*,6,Hangup()
exten = 80*,101,Set(DB(phonelock/${CALLERID(number)})=YES)
exten = 80*,102,Playback(activated)
exten = 80*,103,Hangup()
Doug
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Re: [asterisk-users] Lock extension from asterisk
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
2007/8/17, Gordon Henderson :
S (all untested!)
exten = ,1,Answer()
exten = ,n,Set(me=${CALLERID(num)})
exten = ,n,Set(DB(${me}/locked)=1)
exten = ,1,Answer()
exten = ,n,Set(me=${CALLERID(num)})
exten = ,n,VMAuthenticate(${me})
exten = ,n,Set(DB(${me}/locked)=)
(I think I swapped the and here, but I'm sure you can see that!)
and in the dial-plan where call processing takes place:
exten = s,1,Set(me=${CALLERID(num)})
exten = s,n,Set(locked=${DB(${me}/locked)})
exten = s,n,GotoIf(${locked}?:doneLockCheck)
exten = s,n,Playback(sorry-cant-let-you-do-that)
exten = s,n,Hangup()
exten = s,n(doneLockCheck),Noop(We're not locked)
Works like a charm. Thanks very much.
- --
Andres Jimenez
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Comment: http://firegpg.tuxfamily.org
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W3QsbPV53glyOdxaFVNnFrw=
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[asterisk-users] Connecting a GSM gateway to a FXO port
I am trying to get a GSM gateway (Alpha Tech GSM Gateway Blue Gate Dual
Band Analoog FXO) working with Asterisk.
I had a working FXO configuration to a analog port of a small home 1/4
ISDN pbx.
I used this same configuration to connect a GSM Gateway that is supposed to
be connected to the external(FXO) analog port of a pbx.
I can get my configuration to dial the mobile number via the gateway, but
Asterisk never detects that the remote side (the mobile phone) answered.
When I use the r Dial option it will continue to signal a ringing tone to
the local extension. When I leave out the r, I do get a connection but
only for the duration of the configured number of seconds the Dial command
is waiting for a connection. Because it did not detect the call being
connected it will stop with a NOANSWER status effectively dropping the
connection.
I want to try two possible solutions:
1) I try to find out how to configure the GSM gateway behave in such a way
that Asterisk correctly detects the call being connected.
2) I configure Asterisk to work with this device.
The problem is that I looked thru the configuration options of the gateway
and I could not find anything that made sense to me in relation to this
prb.
What (how) is the gateway supposed to signal back that the call is
connected? If I know what is needed I can go back to my supplier with the
right questions.
If the gateway can not be configured properly, I want to know how I could
configure Asterisk to work around this problem and make it work anyway.
This is the zapata.conf for the FXO port:
[channels]
; hardware channels
; default
usecallerid=yes
hidecallerid=no
callwaiting=no
threewaycalling=yes
transfer=yes
echocancel=yes
echotraining=yes
immediate=no
flash=100
hanguponpolarityswitch=yes
callprogress=yes
progzone=nl
busydetect=yes
busycount=4
language=nl
; define channels
context=binnenkomend
signalling=fxs_ks
channel = 1 ; pstn attached to port 1
This is the Dialplan fragment:
exten = _87.,1,Wait(0.5)
exten = _87.,n,Dial(Zap/1/${EXTEN:2},30,rtT)
exten = _87.,n,Macro(fastbusy)
exten = _87.,n,Hangup
exten = _87.,102,Playback(tt-allbusy)
[macro-fastbusy]
exten = s,1,Answer
exten = s,n,Wait,1
exten = s,n,Playback(vm-isunavail)
exten = s,n,Wait(3)
exten = s,n,Hangup
-- Executing [EMAIL PROTECTED]:1] Wait(SIP/kimura1-08236550, 0.5)
in new stack
-- Executing [EMAIL PROTECTED]:2] Dial(SIP/kimura1-08236550,
Zap/1/0623027714|30|rtT) in new stack
-- Called 1/0623027714
-- Nobody picked up in 3 ms
-- Hungup 'Zap/1-1'
-- Executing [EMAIL PROTECTED]:3] Macro(SIP/kimura1-08236550,
fastbusy) in new stack
-- Executing [EMAIL PROTECTED]:1] Answer(SIP/kimura1-08236550, )
in new stack
-- Executing [EMAIL PROTECTED]:2] Wait(SIP/kimura1-08236550, 1) in
new stack
-- Executing [EMAIL PROTECTED]:3] Playback(SIP/kimura1-08236550,
vm-isunavail) in new stack
-- SIP/kimura1-08236550 Playing 'vm-isunavail' (language 'nl')
-- Executing [EMAIL PROTECTED]:4] Wait(SIP/kimura1-08236550, 3) in
new stack
-- Executing [EMAIL PROTECTED]:5] Hangup(SIP/kimura1-08236550, )
in new stack
== Spawn extension (macro-fastbusy, s, 5) exited non-zero on
'SIP/kimura1-08236550' in macro 'fastbusy'
== Spawn extension (macro-fastbusy, s, 5) exited non-zero on
'SIP/kimura1-08236550'
Any help would be appreciated.
Hans Feringa
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Re: [asterisk-users] 99 bottles of beer
I dialed it, but I am still thirsty. ;-)
On Thu, 2007-08-16 at 19:38 -0600, Steve Murphy wrote:
On Thu, 2007-08-16 at 07:56 -0400, Russell Bryant wrote:
Gordon Henderson wrote:
; *99:
; 99 bottles of beer on the wall.
exten = *99,1,Noop(99 Bottles of beer on the wall)
exten = *99,n,Answer()
exten = *99,n,Set(bottles=99)
exten = *99,n(loop),Noop(There are ${bottles} bottles of beer on
the wall)
exten = *99,n,SayNumber(${bottles})
exten = *99,n,Noop(Take one done and pass it round and there's)
exten = *99,n,Set(bottles=$[${bottles}-1])
exten = *99,n,Noop(${bottles} bottles of beer on the wall)
exten = *99,n,SayNumber(${bottles})
exten = *99,n,GotoIf($[${bottles} 0]?loop)
exten = *99,n,Noop(We're out of beer!)
exten = *99,n,Hangup()
Too much dial plan mashing this morning and I rememberd this site:
http://99-bottles-of-beer.net/
And now, in AEL! (This is untested, I just wanted to see how it would
look.)
context silly {
*99 = {
NoOp(99 Bottles of beer on the wall);
Answer();
bottles=99;
while (${bottles} 0) {
NoOp(${bottles} bottles of beer on the wall, ${bottles} bottles
of beer);
SayNumber(${bottles});
NoOp(Take one down, pass it around);
bottles=${bottles} - 1;
NoOp(${bottles} bottles of beer on the wall);
}
NoOp(We're out of beer!);
Hangup();
}
}
Lol, Well done, Russell!
How about this one: from an extensions.conf that someone posted on the
internet, I think, and I converted to AEL; I'm sorry, but I can't find
the original author.
(If anybody can find his post, I'd love to give him credit.) I did test
this out,
and it works; just put a call to the macro ( guessgame(); ) in an
extension in your dialplan
macro guessgame()
{
startpoint:
while (1)
{
Playback(guessit/intro);
set(GUESS=);
GUESS=${EPOCH}%9;
Set(TIMEOUT(digit)=3);
Set(TIMEOUT(response)=5);
while (1)
{
Read(NUMBER,guessit/input_number,1);
Verbose(Got ${NUMBER} from Read);
if( ${NUMBER} = * || ${NUMBER} = # ||
${NUMBER} = )
{
Playback(guessit/thatsnotanumber);
}
else if (${NUMBER} = ${GUESS})
{
Playback(guessit/win);
break; // the only way out of this loop!
}
else if (${NUMBER} ${GUESS})
{
Playback(guessit/less);
}
else if (${NUMBER} ${GUESS})
{
Playback(guessit/more);
}
else /* what other stuff can the user enter than a
number, #,
nothing, or * ? */
{
Playback(guessit/thatsnotanumber);
}
}
/* You get here after a successful guess */
Wait(.5);
Read(AGAIN,guessit/playagain,1);
if (${AGAIN} != 1)
break;
}
Playback(guessit/goodbye);
return;
catch t
{
playback(guessit/goodbye);
return;
}
catch i
{
playblack(invalid);
}
}
murf
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Hey murf,
here is the link for the credit,
http://www.voipphreak.ca/archives/358-Asterisk-Howto-Numbers-Guessing-Game.html
its also in the wiki examples.
http://www.voip-info.org/wiki/view/AEL+Example+Snippets
db
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Re: [asterisk-users] Lock extension from asterisk
On Fri, 17 Aug 2007, Doug Lytle wrote:
Gordon Henderson wrote:
On Fri, 17 Aug 2007, Andres Jimenez wrote:
exten = ,1,Answer()
exten = ,n,Set(me=${CALLERID(num)})
exten = ,n,Set(DB(${me}/locked)=1)
exten = ,1,Answer()
exten = ,n,Set(me=${CALLERID(num)})
exten = ,n,VMAuthenticate(${me})
exten = ,n,Set(DB(${me}/locked)=)
This is good, but it can be done with just 1 extension:
exten = 80*,1,Answer()
exten = 80*,2,Set(LOCKED=${DB(phonelocked/${CALLERID(number)})})
exten = 80*,3,GotoIf($[${LOCKED} = YES]?80*,4:80*,101)
exten = 80*,4,Set(DB(phonelocked/${CALLERID(number)})=NO)
exten = 80*,5,Playback(de-activated)
exten = 80*,6,Hangup()
exten = 80*,101,Set(DB(phonelock/${CALLERID(number)})=YES)
exten = 80*,102,Playback(activated)
exten = 80*,103,Hangup()
XOR in dialplan :)
9 lines of code vs. my 7 (which include a validation) though ;-)
Dialplan obscurification contest, anyone? (joking - it's weird enough as
it is!!!)
One thing I noted recently is that phones sometimes do weird things with
*'s and #'s )-: The Siemens C460IP DECT phones in particular won't let you
dial a number that has a * in it at any position other than the 1st digit
- I guess this is because the phone uses it to switch from the default
interface to the other one (PSTN vs. VoIP)
Gordon
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Re: [asterisk-users] Paging: Does anyone have a simple howto for Polycoms?
Doug wrote:
I've looked at the following pages, and they are
just so garbled. I keep going around in circles:
http://www.voip-info.org/wiki/view/Polycom+auto-answer+config
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+page
http://www.voip-info.org/wiki/index.php?page=Asterisk+Paging+and+Intercom
http://lists.digium.com/pipermail/asterisk-users/2006-May/152764.html
http://threebit.net/mail-archive/asterisk-users/msg23241.html
http://www.aussievoip.com.au/wiki/freePBX-Paging
http://www.voip-info.org/wiki/view/Setting+up+paging+with+a+sound+card
Can anyone just show some simple working examples on
1. The Asterisk side
2. The Polycom side
Here's what I use on my production system with 1.2.24. This for one of
our four page zones but it happens to page through the polycoms in the
office.
In extensions.conf I have the following:
[pagezones]
; Office page zone through phones
; I don't want to see page calls in my cdr reports.
exten = _631,1,SetAMAFlags(omit)
; There are actually several more phones in here but I cut them
; out for readability
exten = _631,n,Page(Local/[EMAIL PROTECTED]Local/[EMAIL PROTECTED])
exten = _631,n,Hangup
[intercom]
exten = _133XX,1,Macro(pageextension,SIP/${EXTEN:1})
[macro-pageextension]
; Paging macro:
; Check to see if device is in use and DO NOT PAGE if they are
; ${ARG1} - Device to page
;
exten = s,1,ChanIsAvail(${ARG1}|js) ; j for jump and s is for ANY call
exten = s,n,Set(_ALERT_INFO=page) ; This is for the PolyComs
exten = s,n,Dial(${ARG1}||)
exten = s,n,Hangup
exten = s,102,Hangup
The [pagezones] context is included in each phones context to make it
available. What happens is the page zone extension is dialed, the page
app is called with several local channels in the [intercom] context. All
of these channels will be dumped into a meetme conference where everyone
except the person paging is muted. Since it uses meetme you will need a
zaptel timing device or ztdummy loaded. The line in the intercom context
simply calls a macro (which I borrowed from voip-info.org I believe).
The macro first checks to see if the phone is in use. If it is not, then
the _ALERT_INFO header is set to page (more on this below) and the
phone is then dialed. If the phone is in use then that local channel is
hungup and will not be paged to. Paging a phone that is in use causes
some odd things to happen on both the phone and asterisk side sometimes.
On the polycom side, here's what I have set in the sip.cfg (I'm using 1.6.7)
You must fill in values in for the alertInfo tag. It's near the top of
the config file in the voIpProtSIP section. See section 4.6.1.1.3.2
(page 74) of the SIP 1.6 Admin Guide for details. Here is how I filled
mine out:
alertInfo voIpProt.SIP.alertInfo.1.value=page
voIpProt.SIP.alertInfo.1.class=4/
Notice the alertInfo.1.value is set to page, the same as what I set
_ALERT_INFO to in my macro. The class is set to the ring type I want to
use on the phone. RingType is discussed in section 4.6.1.7.2 (page 91).
Mine is set to 4 which corresponds to:
RING_ANSWER se.rt.4.name=Ring Answer se.rt.4.type=ring-answer
se.rt.4.timeout=500 se.rt.4.ringer=13 se.rt.4.callWait=6
se.rt.4.mod=1/
When one of my phones is paged it rings for 1/2 a second and then
automatically answers the incoming call. I have set se.rt.4.ringer=13
because I have created a custom page beep ring tone. You can use one of
the predefined ring tones or if you don't want any page beep set the
ring class in the alertInfo tag to 3 which is auto-answer.
Hope that answers your question.
-Dave
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[asterisk-users] Detecting DTMF Tones from Muted app_meetme Participants
Hi, folks. I have a problem using Asterisk 1.2. I create conferences using app_meetme and Zap channels, and for every participant I run the script defined by AGI_BACKGROUND_SCRIPT to be able to listen and react to DTMF tones. As the docs tell me, when using the AGI background script one loses the ability to control the meetme conference via the command line so for muting conference participants I had to create a small workaround (letting the user leave the conference by letting his AGI background script finish) and then rejoin him using the 'm' flag so that he can keep listening but is not allowed to talk. Now, while the user is muted, I can not read any DTMF tones in the AGI background script. Okay, you might say, the user is muted. It makes sense that his DTMF tones do not come through. But I say, the muting, i.e. the nullification of any incoming audio is done somewhere within Asterisk, why is the DTMF tone detection not done _before_ the incoming audio is thrown away? So, basically, my question is, how can I have a muted participant in a meetme conference that can still control his listening experience via DTMF tones? And I'm not afraid to mess around in the code - somehow, somewhere (at least in my imagination) there have to be two lines, the one reading remove_audio(); and the other one reading detect_dtmf_tones(); which should be rearranged... :) If anyone has a hint on where to start looking in the moloch that is Asterisk, please give me a hint. Or if you know in what other way I can achieve my goal, please give me a hint. I will be grateful until eternity (or next wednesday, whichever comes first). Thanks in advance, David -- INA Service GmbH Papenreye 63 22453 Hamburg Germany Mail: [EMAIL PROTECTED] Phone: +49 (0)40 557 07-07 Fax: +49 (0)40 557 07-100 Geschäftsführer: Ines M. Hoerner Amtsgericht Hamburg HRB 96470 Ust-IdNr. DE248754961 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Lock extension from asterisk
Gordon Henderson wrote: On Fri, 17 Aug 2007, Doug Lytle wrote: XOR in dialplan :) 9 lines of code vs. my 7 (which include a validation) though ;-) Ooops! Missed that line. One thing I noted recently is that phones sometimes do weird things with *'s and #'s )-: The Siemens C460IP DECT phones in particular won't let you Yeah, Polycom's use the # internally as well. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call Limits
Hi all, Some of my asterisk users have used their maximum call limit for incoming calls (peers). There incoming call limit should automatically reset to zero after hangup but its not happening and they no longer can recieve any calls as their allowed limit is already full. So is there any way to reset the call limit on peers by commands or do i have to restart my asterisk server? -- Best Regards Rizwan Hisham Software Engineer Axvoice Inc. www.axvoice.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Connecting a GSM gateway to a FXO port
On Fri, 17 Aug 2007, Hans Feringa wrote:
I am trying to get a GSM gateway (Alpha Tech GSM Gateway Blue Gate Dual
Band Analoog FXO) working with Asterisk.
I had a working FXO configuration to a analog port of a small home 1/4
ISDN pbx.
I used this same configuration to connect a GSM Gateway that is supposed to
be connected to the external(FXO) analog port of a pbx.
I can get my configuration to dial the mobile number via the gateway, but
Asterisk never detects that the remote side (the mobile phone) answered.
You're dialling the number of the SIM card in the gateway? Don't you want
to dial another mobile number?
I have a GSM gateway here - (Telecom FM unit) I connected it into an FXO
port, stuck a SIM card in it, and it just worked. I can route outgoing
calls through it and take incoming calls from it. As far as asterisk is
concerned, it's just another FXO port and it treats it no different to the
FXO port connected to my BT analog line...
Have you tried connecting a normal analogue phone to the unit and seeing
what happens? Can you pick up the phone, get a dial-tone and dial a
number? ... and does the phone ring when you call the SIM card number from
another phone?
Maybe the SIM card you've put in it has a PIN that needs unlocking? (You
did put a SIM card in, right?)
When I use the r Dial option it will continue to signal a ringing tone to
the local extension. When I leave out the r, I do get a connection but
only for the duration of the configured number of seconds the Dial command
is waiting for a connection. Because it did not detect the call being
connected it will stop with a NOANSWER status effectively dropping the
connection.
I want to try two possible solutions:
1) I try to find out how to configure the GSM gateway behave in such a way
that Asterisk correctly detects the call being connected.
2) I configure Asterisk to work with this device.
The problem is that I looked thru the configuration options of the gateway
and I could not find anything that made sense to me in relation to this
prb.
What (how) is the gateway supposed to signal back that the call is
connected? If I know what is needed I can go back to my supplier with the
right questions.
It's an analogue device - there is no way to (normally) send these signals
back. Asterisk assumes the call is connected as soon as it's sent the last
DTMF tone down the line... (AIUI)
If the gateway can not be configured properly, I want to know how I could
configure Asterisk to work around this problem and make it work anyway.
The gateway should act just like an ordinary analogue line from your
telephone exchange, so if you can get asterisk to work with one of those,
then you ought to be able to get it to work with the gateway.
At least thats how my unit works!
This is the zapata.conf for the FXO port:
[channels]
; hardware channels
; default
usecallerid=yes
hidecallerid=no
callwaiting=no
threewaycalling=yes
transfer=yes
echocancel=yes
echotraining=yes
immediate=no
flash=100
hanguponpolarityswitch=yes
Why? Does the device explicitly do this to signal a hangup?
callprogress=yes
progzone=nl
busydetect=yes
busycount=4
language=nl
; define channels
context=binnenkomend
signalling=fxs_ks
channel = 1 ; pstn attached to port 1
This is the Dialplan fragment:
exten = _87.,1,Wait(0.5)
exten = _87.,n,Dial(Zap/1/${EXTEN:2},30,rtT)
exten = _87.,n,Macro(fastbusy)
exten = _87.,n,Hangup
exten = _87.,102,Playback(tt-allbusy)
[macro-fastbusy]
exten = s,1,Answer
exten = s,n,Wait,1
exten = s,n,Playback(vm-isunavail)
exten = s,n,Wait(3)
exten = s,n,Hangup
-- Executing [EMAIL PROTECTED]:1] Wait(SIP/kimura1-08236550, 0.5)
in new stack
-- Executing [EMAIL PROTECTED]:2] Dial(SIP/kimura1-08236550,
Zap/1/0623027714|30|rtT) in new stack
-- Called 1/0623027714
-- Nobody picked up in 3 ms
-- Hungup 'Zap/1-1'
-- Executing [EMAIL PROTECTED]:3] Macro(SIP/kimura1-08236550,
fastbusy) in new stack
-- Executing [EMAIL PROTECTED]:1] Answer(SIP/kimura1-08236550, )
in new stack
-- Executing [EMAIL PROTECTED]:2] Wait(SIP/kimura1-08236550, 1) in
new stack
-- Executing [EMAIL PROTECTED]:3] Playback(SIP/kimura1-08236550,
vm-isunavail) in new stack
-- SIP/kimura1-08236550 Playing 'vm-isunavail' (language 'nl')
-- Executing [EMAIL PROTECTED]:4] Wait(SIP/kimura1-08236550, 3) in
new stack
-- Executing [EMAIL PROTECTED]:5] Hangup(SIP/kimura1-08236550, )
in new stack
== Spawn extension (macro-fastbusy, s, 5) exited non-zero on
'SIP/kimura1-08236550' in macro 'fastbusy'
== Spawn extension (macro-fastbusy, s, 5) exited non-zero on
'SIP/kimura1-08236550'
Any help would be appreciated.
Hans Feringa
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Re: [asterisk-users] A102 card, BT ISDN30e, silence
In article [EMAIL PROTECTED],
Rory Campbell-Lange [EMAIL PROTECTED] wrote:
Summary:
Can't hear incoming/outgoing calls to/from ISDN over Sangoma A102 card.
Debian stable + Asterisk 1:1.2.13.
Thanks for the response, Andres. We've changed the timing source but
still no joy. This is very odd; calling between internal AIX/SIP
extensions works fine.
If anyone else can think of anything, I would be grateful to hear.
Try setting echocancel=no instead of yes in zaptel.conf.
Also, are you able to try a different, known working system on the same
telco ISDN port (just in case the problem is not in your system after all).
Do you have two systems, or two E1 ports on a single system? If so, you
could connect them to each other using a crossover cable (1,2)-(4,5)
and place calls from one to the other. One would need to be set to pri_cpe
and the other to pri_net, but Asterisk will tell you if you forget.
Cheers
Tony
Rory
On 16/08/07, Andres Paglayan ([EMAIL PROTECTED]) wrote:
On Aug 16, 2007, at 8:58 AM, Rory Campbell-Lange wrote:
However both incoming and outgoing calls are greeted by silence.
...
Zaptel
--
loadzone=uk
defaultzone=uk
#Sangoma A101 port 1 [slot:3 bus:2 span: 1]
span=1,0,0,ccs,hdb3,crc4
the first 0 is the timing source,
if you are connecting this span to a telco
it should be 1 for your telco to provide timing,
which is what they expect
; note that only 25 channels are active
bchan=1-15,17-31
dchan=16
Zapata
--
-
[channels]
language=en
context=from-pstn
signalling=pri_cpe
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=no
echotraining=800
rxgain=0.0
txgain=0.0
group=0
callgroup=1
pickupgroup=1
immediate=no
pridialplan=unknown
prilocaldialplan=unknown
group=1
callerid=asreceived
;Sangoma A101 port 1 [slot:3 bus:2 span: 1]
switchtype=euroisdn
context=from-pstn
group=0
signalling=pri_cpe
channel = 1-15,17-31
Extensions
--
-
[general]
static=yes
writeprotect=no
autofallthrough=yes
clearglobalvars=no
priorityjumping=no
[globals]
CONSOLE=Console/dsp ; Console interface for demo
IAXINFO=guest ; IAXtel username/password
TRUNKMSD=1; MSD digits to strip (usually 1 or 0)
[dundi-e164-local]
switch = DUNDi/e164
[local]
ignorepat = 9
include = default
include = parkedcalls
[default]
exten = 001,1,Dial(SIP/001,5)
exten = 001,n,VoiceMail([EMAIL PROTECTED])
exten = 001,n,Playback(vm-goodbye)
exten = 001,n,Hangup
exten = 901,1,Dial(IAX2/901,5)
exten = 901,n,Hangup
exten = 100,1,Dial(SIP/100,5)
exten = 100,n,VoiceMail([EMAIL PROTECTED])
exten = 100,n,Playback(vm-goodbye)
exten = 100,n,Hangup
exten = _0XX,1,Dial(Zap/g0/${EXTEN})
exten = _ZX,1,Dial(Zap/g0/${EXTEN})
[from-pstn]
; BT pass in 6 digits only?
exten = 032685,1,Answer()
exten = 032685,2,Playback(demo-echotest)
exten = 032685,3,Echo()
exten = 032685,4,Playback(demo-echodone)
exten = 032685,5,Hangup()
--
Rory Campbell-Lange
Director
Campbell-Lange Workshop Ltd.
[EMAIL PROTECTED]
www.campbell-lange.net
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Andres Paglayan
--Harmony is more important than being right
Bapak
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--
Rory Campbell-Lange
Director
Campbell-Lange Workshop Ltd.
[EMAIL PROTECTED]
www.campbell-lange.net
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--
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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Re: [asterisk-users] Lock extension from asterisk
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Thanks to Gordon Doug I have now a very good locking system using
one only extension.
Extension informs you about the current lock situation and, if
authentacated, it changes it and explain the change done.
;Locking system
;LOCK
exten = ,1,Answer()
exten = ,n,Set(me=${CALLERID(num)})
exten = ,n,GotoIf(${DB(${me}/locked)}?,101:,201)
exten = ,101,Playback(security)
exten = ,n,Playback(activated)
exten = ,n,VMAuthenticate(${me})
exten = ,n,Set(DB(${me}/locked)=)
exten = ,n,Playback(security)
exten = ,n,Playback(now)
exten = ,n,Playback(de-activated)
exten = ,n,Hangup()
exten = ,201,Playback(security)
exten = ,n,Playback(de-activated)
exten = ,n,VMAuthenticate(${me})
exten = ,n,Set(DB(${me}/locked)=1)
exten = ,n,Playback(security)
exten = ,n,Playback(now)
exten = ,n,Playback(activated)
exten = ,n,Hangup()
Thanks again, guys
- --
Andres Jimenez
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.3 (GNU/Linux)
Comment: http://firegpg.tuxfamily.org
iD8DBQFGxasI8SZxpGYWwpYRAsMZAJwPb/fRAH5IMB4muBtzH1QIPfMFlgCeI2iu
UZH/bs38f1iqiZ/CNWvoTsk=
=Fsal
-END PGP SIGNATURE-
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Re: [asterisk-users] No audio on ISDN PRI calls
Get the providers support on the line and I will bet anything you are not hitting their side of the line, this is most likely a signaling or channel numbering issue. Anthony Veselin Kantsev wrote: Hello, I have a Sangoma A101 connected to an ISDN30 (E1 in the UK) with some Snom 300 and Idefisk softphones. I can do SIP and IAX2 calls just fine, however I cant get any audio in either direction on the Zap channels. When I call in or dial out over the ISDN30 (UK E1) I can see the call answered/placed on the CLI and then silence follows. I've been provisioned 25 out of the 31 channels only at the moment and I'm using a dummy number given by BT for testing. The is no alarms on the line, nor errors on the CLI or in the /va/log/messages or wanrouter. Any ideas? Thank you much. Veselin ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem in installing libmfcr2 for configuring MFC/R2
I think you did not installed properly the libraries. libmfcr2 check for: checking tiffio.h usability... yes checking tiffio.h presence... yes checking for tiffio.h... yes checking for TIFFOpen in -ltiff... yes That is, you need to have BOTH, headers and shared library. ldd in protocol_mfcr2.so shows: libtiff.so.3 = /usr/lib/libtiff.so.3 Where do you have libtiff installed? On 8/17/07, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Fri, Aug 17, 2007 at 01:23:11PM +0530, [EMAIL PROTECTED] wrote: Hi, I am trying to configure for MFC/R2 for asterisk. With the help of one of the asterisk users group member patrick I am able to install libunicall library. Now, when trying to install libmfr2-0.0.3 it is giving error. On running running command $./configure It is giving error - can't build without libtiff I downloaded tiff-3.8.2 and installed the libraries but still the problem persists. Can anybody will tell me how to remove the problem. Thanx and regards sanchal Look in config.log (near the end). What command fails that causes this error? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Within C++, there is a much smaller and cleaner language struggling to get out. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Limits
Have you looked in show channels/core show channels to see if they have any dead/zombie channels, which you can remove with soft-hangup? What version of * are you running? What kind of phones? What config options are you using in SIP (or other tech) to limit the calls? On 8/17/07, Rizwan Hisham [EMAIL PROTECTED] wrote: Hi all, Some of my asterisk users have used their maximum call limit for incoming calls (peers). There incoming call limit should automatically reset to zero after hangup but its not happening and they no longer can recieve any calls as their allowed limit is already full. So is there any way to reset the call limit on peers by commands or do i have to restart my asterisk server? -- Best Regards Rizwan Hisham Software Engineer Axvoice Inc. www.axvoice.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Anthony Cennami ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Connecting a GSM gateway to a FXO port
Thanks for your response. My answers below.
On Fri, 17 Aug 2007, Hans Feringa wrote:
I am trying to get a GSM gateway (Alpha Tech GSM Gateway Blue Gate Dual
Band Analoog FXO) working with Asterisk.
I had a working FXO configuration to a analog port of a small home 1/4
ISDN pbx.
I used this same configuration to connect a GSM Gateway that is supposed
to
be connected to the external(FXO) analog port of a pbx.
I can get my configuration to dial the mobile number via the gateway,
but
Asterisk never detects that the remote side (the mobile phone) answered.
You're dialling the number of the SIM card in the gateway? Don't you want
to dial another mobile number?
No I am dialing out to another mobile number (bad english on my part).
I have a GSM gateway here - (Telecom FM unit) I connected it into an FXO
port, stuck a SIM card in it, and it just worked. I can route outgoing
calls through it and take incoming calls from it. As far as asterisk is
concerned, it's just another FXO port and it treats it no different to the
FXO port connected to my BT analog line...
Have you tried connecting a normal analogue phone to the unit and seeing
what happens? Can you pick up the phone, get a dial-tone and dial a
number? ... and does the phone ring when you call the SIM card number from
another phone?
Works with an analogue phone. ..
Dialing out from Asterisk to a mobile number works, however picking up the
mobile phone and establishing the connection is not detected by Asterisk.
(see below)
Maybe the SIM card you've put in it has a PIN that needs unlocking? (You
did put a SIM card in, right?)
No pin required ..
When I use the r Dial option it will continue to signal a ringing tone
to
the local extension. When I leave out the r, I do get a connection but
only for the duration of the configured number of seconds the Dial
command
is waiting for a connection. Because it did not detect the call being
connected it will stop with a NOANSWER status effectively dropping the
connection.
I want to try two possible solutions:
1) I try to find out how to configure the GSM gateway behave in such a
way
that Asterisk correctly detects the call being connected.
2) I configure Asterisk to work with this device.
The problem is that I looked thru the configuration options of the
gateway
and I could not find anything that made sense to me in relation to this
prb.
What (how) is the gateway supposed to signal back that the call is
connected? If I know what is needed I can go back to my supplier with
the
right questions.
It's an analogue device - there is no way to (normally) send these signals
back. Asterisk assumes the call is connected as soon as it's sent the last
DTMF tone down the line... (AIUI)
That is what I understood at first. I could not understand it's different
behaviour from the previous connection to the pbx. Maybe it has something
to do with the current that flows when it is off hook?
If the gateway can not be configured properly, I want to know how I
could
configure Asterisk to work around this problem and make it work anyway.
The gateway should act just like an ordinary analogue line from your
telephone exchange, so if you can get asterisk to work with one of those,
then you ought to be able to get it to work with the gateway.
My thoughts exactly.
At least thats how my unit works!
This is the zapata.conf for the FXO port:
[channels]
; hardware channels
; default
usecallerid=yes
hidecallerid=no
callwaiting=no
threewaycalling=yes
transfer=yes
echocancel=yes
echotraining=yes
immediate=no
flash=100
hanguponpolarityswitch=yes
Why? Does the device explicitly do this to signal a hangup?
Hmm, a leftover of my testing the hangup situation on the pbx. I guess I
should remove that, because it started working by configuring busydetect.
callprogress=yes
progzone=nl
busydetect=yes
busycount=4
language=nl
; define channels
context=binnenkomend
signalling=fxs_ks
channel = 1 ; pstn attached to port 1
This is the Dialplan fragment:
exten = _87.,1,Wait(0.5)
exten = _87.,n,Dial(Zap/1/${EXTEN:2},30,rtT)
exten = _87.,n,Macro(fastbusy)
exten = _87.,n,Hangup
exten = _87.,102,Playback(tt-allbusy)
[macro-fastbusy]
exten = s,1,Answer
exten = s,n,Wait,1
exten = s,n,Playback(vm-isunavail)
exten = s,n,Wait(3)
exten = s,n,Hangup
-- Executing [EMAIL PROTECTED]:1] Wait(SIP/kimura1-08236550,
0.5)
in new stack
-- Executing [EMAIL PROTECTED]:2] Dial(SIP/kimura1-08236550,
Zap/1/0623027714|30|rtT) in new stack
-- Called 1/0623027714
-- Nobody picked up in 3 ms
-- Hungup 'Zap/1-1'
-- Executing [EMAIL PROTECTED]:3] Macro(SIP/kimura1-08236550,
fastbusy) in new stack
-- Executing [EMAIL PROTECTED]:1] Answer(SIP/kimura1-08236550, )
in new stack
-- Executing [EMAIL PROTECTED]:2] Wait(SIP/kimura1-08236550, 1)
in
new stack
-- Executing [EMAIL PROTECTED]:3] Playback(SIP/kimura1-08236550,
vm-isunavail) in new stack
[asterisk-users] DISA and Ericsson Dialog 3212
Hello fellows!!! I'm having problems with Ericsson Dialog 3221 phone and DISA. I've configured an extension to test DISA and it work properly with all other phones, but freeze with the mentioned phone. Here is my extension: exten = 105,1,Answer exten = 105,2,Set(TIMEOUT(digit)tting =5) exten = 105,3,Set(TIMEOUT(response)=10) exten = 105,4,DISA(no-password|default) Asterisk 1.4.10 / Debian Etch Thanks for any help! Regards, McCoy ** ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Jain-Sip-Applet-Phone with Asterisk
Hello, I have the Jain-Sip-Applet-Phone installed on two machines in a small LAN network. These machines are connected through an Asterisk Server (Using Trixbox). I run the phone as an application on both machines through Eclipse and I am able to log on as a user with one of the extensions that I use within Asterisk on each machine (extensions 201 and 202 in this case). When I try to add a contact to my list (the other user machine) from one of the machines, I get the following message, which looks like I am having a problem with authentication: ECLIPSE CONSOLE WINDOW ![CDATA[SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.252:7933;branch=z9hG4bKf87a334e3afb041a4b7783d409b6c95d;received=192.168.1.252 From: sip:[EMAIL PROTECTED];tag=973 To: sip:[EMAIL PROTECTED];tag=as72bce758 Call-ID: [EMAIL PROTECTED] CSeq: 1 SUBSCRIBE User-Agent: Asterisk PBX Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY WWW-Authenticate: Digest algorithm=MD5,realm=asterisk,nonce=62a12192 Content-Length: 0 I have defined two users in my sip.conf file as shown below: ; do not edit this file, this is an auto-generated file by freepbx ; all modifications must be done from the web gui [201] type=friend secret=201 record_out=Adhoc record_in=Adhoc qualify=yes port=5060 nat=yes host=dynamic dtmfmode=rfc2833 dial=SIP/201 context=from-internal canreinvite=no callerid=device 201 [202] type=friend secret=202 record_out=Adhoc record_in=Adhoc qualify=yes port=5060 nat=yes host=dynamic dtmfmode=rfc2833 dial=SIP/202 context=from-internal canreinvite=no callerid=device 202 Would anyone know where I am going wrong? Is it possible to remove this authentication requirement? Thanks in advance, Denis Kutman ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Limits
Thanx for ur reply. Im running * 1.4.2. i dont think there is any problem in asterisk because only one user is having this problem. User is using Aastra 480i Cordless phone Here is the sip config for that user. Im using call-limit=2 for every user [saadfarr] username=saadfarr type=friend secret=123 qualify=no nat=yes insecure=port,invite call-limit=2 host=dynamic dtmfmode=rfc2833 context=local canreinvite=yes accountcode=1:0:saadfarr amaflags=default sip show channels give me the following: Peer User/ANRCall ID Seq (Tx/Rx) Form Hold Last Message 68.144.208.153 (None) 1aed3833095 00101/00102 unkn No Rx: OPTIONS 85.234.144.137 (None) 256a1914164 00101/01429 unkn No Rx: REGISTER 216.143.130.70 1812923551 70f46ee82be 00102/0 unkn No Tx: ACK 74.96.225.223saadfarr0c666dc33b3 00102/0 unkn No Init: INVITE 74.96.225.223saadfarr522a18fd48c 00102/0 unkn No Init: INVITE 66.131.246.220 foahand24e8cfe9c416 00102/0 unkn No Init: INVITE 124.29.216.185 1212933903 443fdaeb50c 00102/0 unkn No Init: INVITE 7 active SIP channels How do i know which one is dead/zombie channel. I can see 2 channels for user saadfarr. i tried to use soft hangup but it requires channel name...how do i know the channel name if its a zombie channel. On 8/17/07, Anthony Cennami [EMAIL PROTECTED] wrote: Have you looked in show channels/core show channels to see if they have any dead/zombie channels, which you can remove with soft-hangup? What version of * are you running? What kind of phones? What config options are you using in SIP (or other tech) to limit the calls? On 8/17/07, Rizwan Hisham [EMAIL PROTECTED] wrote: Hi all, Some of my asterisk users have used their maximum call limit for incoming calls (peers). There incoming call limit should automatically reset to zero after hangup but its not happening and they no longer can recieve any calls as their allowed limit is already full. So is there any way to reset the call limit on peers by commands or do i have to restart my asterisk server? -- Best Regards Rizwan Hisham Software Engineer Axvoice Inc. www.axvoice.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Anthony Cennami ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Rizwan Hisham Software Engineer Axvoice Inc. www.axvoice.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Suggestions on how to debug strange DTMF problems
I'm hoping people can suggest some ideas for debugging a problem that I'm having with DTMF. Unlike most of the DTMF problems reported here, it has nothing to do with Asterisk interpreting DTMF. My problem is with the synthesis of DTMF tones on outbound calls on a PRI connected to a TE412P card. I'm running * 1.4.10.1 with Zaptel 1.4.4. It is important to note that these problems never occurred when we were using 1.2. They started immediately after upgrading to 1.4 (1.4.4 and 1.4.2.1 at the time IIRC) The original problem people would report is that they have problems using IVRs that they call over the PSTN. Duplicated digits, missed digits, etc. I've tried to replicate the problem using various user agents (Polycom 430, Grandstream GXP2000, Linksys SPA942, eyeBeam, etc.) and have observed the problem. It's not consistent, but that's probably due to the inconsistency of user input. If you concentrate very carefully and hit the keys for a consistent period of time with consistent spacing, the problem doesn't seem to happen. I also found that when the user agent was directed into DISA() and then dialed the IVR from within that application, they didn't have problems with their DTMF being recognized. In an attempt to quantify the problem, I set up the following test harness, sending calls out of Asterisk into an ITSP system. The final termination of the call is a different Asterisk box from the one that generated the call, though running the same version. DTMF from the ITSP to the second Asterisk system is SIP INFO. Asterisk - PRI (TE412P) - PSTN - ITSP - Asterisk I generated calls that executed SendDTMF(). On the other Asterisk system, I used Read() to capture the DTMF and stick it in a database. I did 1250 calls (250 to each of the 4 PRIs connected to the TE412P and 250 without regard to the PRI used). Across all of those calls, I didn't see one missed or doubled-up digit. I then did some manual tests where I placed calls from the Polycom 430 to the same application (basically putting myself in place of SendDTMF). Immediately I saw doubled up an missed digits. I turned on DTMF debugging on the second Asterisk box, and doubled-up digits always seem to take this form: [Aug 17 10:50:05] DTMF[25160] channel.c: DTMF begin emulation of '4' with duration 90 queued on SIP/5060-08da0d70 [Aug 17 10:50:05] DTMF[25160] channel.c: DTMF end emulation of '4' queued on SIP/5060-08da0d70 [Aug 17 10:50:05] DTMF[25160] channel.c: DTMF end '4' received on SIP/5060-08da0d70, duration 70 ms [Aug 17 10:50:05] DTMF[25160] channel.c: DTMF begin emulation of '4' with duration 80 queued on SIP/5060-08da0d70 [Aug 17 10:50:05] DTMF[25160] channel.c: DTMF end emulation of '4' queued on SIP/5060-08da0d70 [Aug 17 10:50:05] DTMF[25160] channel.c: DTMF end '4' received on SIP/5060-08da0d70, duration 50 ms [Aug 17 10:50:05] DTMF[25160] channel.c: DTMF begin emulation of '4' with duration 80 queued on SIP/5060-08da0d70 [Aug 17 10:50:05] DTMF[25160] channel.c: DTMF end emulation of '4' queued on SIP/5060-08da0d70 That's a single keypress. As I understand it, synthesized DTMF tones should consistently have a duration of 100 ms, and indeed most of what shows up in the DTMF log does have that duration (or something close to it): [Aug 17 10:49:55] DTMF[25160] channel.c: DTMF end '1' received on SIP/5060-08da0d70, duration 100 ms [Aug 17 10:49:55] DTMF[25160] channel.c: DTMF begin emulation of '1' with duration 100 queued on SIP/5060-08da0d70 [Aug 17 10:49:55] DTMF[25160] channel.c: DTMF end emulation of '1' queued on SIP/5060-08da0d70 [Aug 17 10:49:57] DTMF[25160] channel.c: DTMF end '2' received on SIP/5060-08da0d70, duration 100 ms [Aug 17 10:49:57] DTMF[25160] channel.c: DTMF begin emulation of '2' with duration 100 queued on SIP/5060-08da0d70 [Aug 17 10:49:57] DTMF[25160] channel.c: DTMF end emulation of '2' queued on SIP/5060-08da0d70 [Aug 17 10:49:58] DTMF[25160] channel.c: DTMF end '3' received on SIP/5060-08da0d70, duration 110 ms [Aug 17 10:49:58] DTMF[25160] channel.c: DTMF begin emulation of '3' with duration 110 queued on SIP/5060-08da0d70 [Aug 17 10:49:58] DTMF[25160] channel.c: DTMF end emulation of '3' queued on SIP/5060-08da0d70 [Aug 17 10:50:00] DTMF[25160] channel.c: DTMF end '4' received on SIP/5060-08da0d70, duration 100 ms [Aug 17 10:50:00] DTMF[25160] channel.c: DTMF begin emulation of '4' with duration 100 queued on SIP/5060-08da0d70 [Aug 17 10:50:01] DTMF[25160] channel.c: DTMF end emulation of '4' queued on SIP/5060-08da0d70 [Aug 17 10:50:02] DTMF[25160] channel.c: DTMF end '4' received on SIP/5060-08da0d70, duration 100 ms Yet from time to time, the ITSP hears that supposed 100ms tone as two tones of 70 and 50ms, or 90 and 40, or some combination that makes it appear to be a doubled-up digit. I'd be tempted to say that the problem is whatever part of the ITSP that is interpreting the DTMF coming in from the PSTN, but the IVRs that my users complain about are operated by tons of different
Re: [asterisk-users] Call Limits
If I recall 1.4.2 has a deadlock problem on their SIP channels -- you should upgrade to at least 1.4.5, which is when this was resolved. The problem was present in 1.2 and 1.4 -- 1.4.5 earliest that I believe does not have this issue. Anthony On 8/17/07, Rizwan Hisham [EMAIL PROTECTED] wrote: Thanx for ur reply. Im running * 1.4.2. i dont think there is any problem in asterisk because only one user is having this problem. User is using Aastra 480i Cordless phone Here is the sip config for that user. Im using call-limit=2 for every user [saadfarr] username=saadfarr type=friend secret=123 qualify=no nat=yes insecure=port,invite call-limit=2 host=dynamic dtmfmode=rfc2833 context=local canreinvite=yes accountcode=1:0:saadfarr amaflags=default sip show channels give me the following: Peer User/ANRCall ID Seq (Tx/Rx) Form Hold Last Message 68.144.208.153 (None) 1aed3833095 00101/00102 unkn No Rx: OPTIONS 85.234.144.137 (None) 256a1914164 00101/01429 unkn No Rx: REGISTER 216.143.130.70 1812923551 70f46ee82be 00102/0 unkn No Tx: ACK 74.96.225.223saadfarr0c666dc33b3 00102/0 unkn No Init: INVITE 74.96.225.223saadfarr522a18fd48c 00102/0 unkn No Init: INVITE 66.131.246.220 foahand24e8cfe9c416 00102/0 unkn No Init: INVITE 124.29.216.185 1212933903 443fdaeb50c 00102/0 unkn No Init: INVITE 7 active SIP channels How do i know which one is dead/zombie channel. I can see 2 channels for user saadfarr. i tried to use soft hangup but it requires channel name...how do i know the channel name if its a zombie channel. On 8/17/07, Anthony Cennami [EMAIL PROTECTED] wrote: Have you looked in show channels/core show channels to see if they have any dead/zombie channels, which you can remove with soft-hangup? What version of * are you running? What kind of phones? What config options are you using in SIP (or other tech) to limit the calls? On 8/17/07, Rizwan Hisham [EMAIL PROTECTED] wrote: Hi all, Some of my asterisk users have used their maximum call limit for incoming calls (peers). There incoming call limit should automatically reset to zero after hangup but its not happening and they no longer can recieve any calls as their allowed limit is already full. So is there any way to reset the call limit on peers by commands or do i have to restart my asterisk server? -- Best Regards Rizwan Hisham Software Engineer Axvoice Inc. www.axvoice.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Anthony Cennami ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Rizwan Hisham Software Engineer Axvoice Inc. www.axvoice.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Anthony Cennami ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Patent issues, what features we can't use?
On Thu, Aug 16, 2007 at 11:00:05PM -0400, Zeeshan Zakaria wrote: This is really ridiculous. So this means that now nobody can use fax-to-email without paying to J2 first? Horseshit. Prior art is trivial. How old is Hylafax? Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dialplan / AGI autoanswer question
Thanks, I'll take a look at it and see what new tricks I can learn. I did
use the astdb initally in my first version but at the time didn't see a good
way to add the state field that I wanted in there. At the moment I have a
number of external programs which use the postgres database - is it possible
to connect to the astdb from an external program?
In local.asterisk.users, you wrote:
On Wed, 15 Aug 2007, Matthew Harrell wrote:
The intent of this sequence is to take the incoming callerid, replace it if
known with something in the database, and branch on the state from the DB
and time of the day.
FWIW: I do something similar, but purely in dial-plan using the astdb -
here's an extract, it demos jumping to named labels too:
exten = incoming,1,Noop(New incoming call. CallerId is ${CALLERID(all)})
; See if we have a name:
exten = incoming,n,GotoIf($[${CALLERID(name)} != ]?gotName)
; OK. No Name. Set a default name
exten = incoming,n,Set(CALLERID(name)=Unknown)
exten = incoming,n(gotName),Noop(Carrying on after name check)
; See if we have a number:
exten = incoming,n,GotoIf($[${CALLERID(number)} != ]?gotNumber)
; OK. No Number. Set a default number
exten = incoming,n,Set(CALLERID(number)=Withheld)
exten = incoming,n(gotNumber),Noop(Carrying on after number check)
; Now see if the number is the our internal database which will override any
; name we might have.
exten = incoming,n,Set(name=${DB(cid/${CALLERID(number)})})
exten = incoming,n,GotoIf($[${name} = ]?doneCIDprocessing)
exten = incoming,n,Set(CALLERID(name)=${name})
exten = incoming,n,Noop(We set our name to ${name} from the database)
exten = incoming,n(doneCIDprocessing),Noop(Done with incoming CID processing
- we have a call from ${CALLERID(all)})
You could keep a separate list of number states in the database too, and
extract this. Eg. above the line where we get the name out of the astdb
above:
exten = incoming,n,Set(CALLSTATE=${DB(state/${CALLERID(number)})})
and so on...
I'm not sure what (if any!) benefit this might have over running an
external PHP application... I'd like to think it might actually be quicker
for simple cases like this, but I've never benchmarked it.
Gordon
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Bit Twiddlers, Inc. stamp collecting.
[EMAIL PROTECTED]
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Re: [asterisk-users] Dialplan / AGI autoanswer question
Interesting. I have essentially the same settings but if I don't wait a brief period then I don't get the callerid filled in. This was with 1.2 or 1.4 Haven't tried with new 1.4 branch of Asterisk/Zaptel, but in 1.2, if correctly configured, there was no need to Wait(x) to let zaptel to get the CID on analog lines: it was zaptel itself to not let the call go through the dialplan until the second ring. I think it shoud be something like: usecallerid=yes callerid=asreceived in zapata.conf for relevant channels. Hope this helps... Regards picciux 2007/8/16, Matthew Harrell [EMAIL PROTECTED]: Thanks. I was hoping there might be a way to detect whether the CID routine was done or not. I've still seen occasions where it wasn't available for callers that I know had it. Maybe my phone service is just a little slow sometimes Wait(2) is what I do. Matthew Harrell wrote: First, it seems I have to have a 2 - 3 second wait before the AGI call in order to get valid CID data. Usually 2 seconds suffices for this one setup but during that time the caller has had two rings before the local extension has even begun to ring. Is there something I am doing wrong that causes it to take so long to get the CID? CallerID info is sent between the first and second ring. Well that would explain that problem, wouldn't it? Is there a proper way to wait for the CID data to be filled in if available or is Wait(2) my best option? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Matthew Harrell Never underestimate the power of Bit Twiddlers, Inc. very stupid people in large groups. [EMAIL PROTECTED] ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --=_Part_138095_28898178.1187347107635 Content-Type: text/html; charset=ISO-8859-1 Content-Transfer-Encoding: 7bit Content-Disposition: inline Haven#39;t tried with new 1.4 branch of Asterisk/Zaptel, but in 1.2, if correctly configured, there was no need to quot;Wait(x)quot; to let zaptel to get the CID on analog lines: it was zaptel itself to not let the call go through the dialplan until the second ring. I think it shoud be something like: brbrusecallerid=yesbrcallerid=asreceivedbrbrin zapata.conf for relevant channels.brbrHope this helps...brbrRegardsbrbrpicciuxbrbrdivspan class=gmail_quote2007/8/16, Matthew Harrell lt;a href=mailto:[EMAIL PROTECTED] [EMAIL PROTECTED]/agt;:/spanblockquote class=gmail_quote style=border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;brThanks.nbsp;nbsp;I was hoping there might be a way to detect whether the CID brroutine was done or not.nbsp;nbsp;I#39;ve still seen occasions where it wasn#39;t availablebrfor callers that I know had it.nbsp;nbsp;Maybe my phone service is just a littlebrslow sometimesbrbrbrgt; Wait(2) is what I do. brgt;brgt; Matthew Harrell wrote:brgt;gt;gt;gt; First, it seems I have to have a 2 - 3 second wait before the AGI call inbrgt;gt;gt;gt; order to get valid CID data.nbsp;nbsp;Usually 2 seconds suffices for this one setup brgt;gt;gt;gt; but during that time the caller has had two rings before the local extensionbrgt;gt;gt;gt; has even begun to ring.nbsp;nbsp;Is there something I am doing wrong that causes itbrgt;gt;gt;gt; to take so long to get the CID? brgt;gt;gt; CallerID info is sent between the first and second ring.brgt;gt;brgt;gt; Well that would explain that problem, wouldn#39;t it?nbsp;nbsp;Is there a proper waybrgt;gt; to wait for the CID data to be filled in if available or is Wait(2) my best brgt;gt; option?brgt;gt;brgt;brgt;brgt; ___brgt; --Bandwidth and Colocation Provided by a href=http://www.api-digital.com--;http://www.api-digital.com--/a brgt;brgt; asterisk-users mailing listbrgt; To UNSUBSCRIBE or update options visit:brgt;nbsp;nbsp;nbsp;nbsp;a href=http://lists.digium.com/mailman/listinfo/asterisk-users;http://lists.digium.com/mailman/listinfo/asterisk-users /abrbr--brnbsp;nbsp;Matthew Harrellnbsp;nbsp;nbsp;nbsp;nbsp;nbsp;nbsp;nbsp;nbsp;nbsp;nbsp;nbsp;nbsp;nbsp;nbsp;nbsp;nbsp;nbsp;nbsp;nbsp;nbsp;nbsp;nbsp;nbsp;nbsp;nbsp;Never underestimate the power ofbrnbsp;nbsp;Bit Twiddlers, Inc.nbsp;nbsp;nbsp;nbsp;nbsp;nbsp;nbsp;nbsp;nbsp;nbsp;nbsp;nbsp;nbsp;nbsp;nbsp;nbsp;nbsp;nbsp;nbsp;nbsp;nbsp;nbsp; very stupid people in large groups.brnbsp;nbsp;a href=mailto:[EMAIL PROTECTED] [EMAIL PROTECTED]/abrbr___br--Bandwidth
[asterisk-users] analog lines running agi on hangup question
I have the following dialplan. Everything seems good except for one thing. If the background message is playing and the user hangs up and does not press a digit how do I run an agi on that event. I tried an exten = h,1,agi(smvoice,-digium_failed) but that was never called. I am using 1.4.10 thanks, Jerry --- [smvoice-analog] exten = s,1,Wait(1) exten = s,2,Set(TIMEOUT(absolute)=30) exten = s,3,Background(SM_PRESS_ONE_TO_HEAR_MESSAGE) exten = s,4,Goto(s,3) ; Accept any digit to continue (turn off the timer) exten = _X,1,Set(TIMEOUT(absolute)=0) exten = _X,2,agi(smvoice,-digium_asterisk) exten = i,1,agi(smvoice,-digium_failed) exten = t,1,agi(smvoice,-digium_failed) exten = T,1,agi(smvoice,-digium_failed) exten = failed,1,agi(smvoice,-digium_failed) ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Connecting a GSM gateway to a FXO port
On Fri, 17 Aug 2007, Hans Feringa wrote:
Thanks for your response. My answers below.
exten = _87.,n,Dial(Zap/1/${EXTEN:2},30,rtT)
My only other suggestion would be to remove the timeout...
Other than that there's no major difference between my setup and yours.
For outbound dialling, I use:
exten = s,n,Dial(Zap/G1/${ARG1},,WTon)
I'd drop the 'r' flag you're using - it might mask any tones you might get
back from the PSTN/GSM box.
Gordon
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[asterisk-users] MOH being activated in the middle of a call
Hi, I am using Asterisk 1.2.19 and have noticed a strange behaviour in my system. Sometimes (I could not reproduce it yet), when there is a call in place between one extension and a PSTN number, the MOH suddenly starts to play for one of them, while the other can still hear what is being said in the conversation. When this issue happens, the call needs to be dropped, because the MOH plays forever. Is this a known issue? Any idea about how to solve/avoid it? Thanks in advance ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [asterisk-biz] Skype Outage Leaves Millions Speechless
On Fri, Aug 17, 2007 at 12:51:35PM -0400, Matthew Rubenstein wrote: eBay is that marketplace, owns Skype, that telco, owns PayPal, that bank. This outage should be screaming from the headlines. As those three essential services become essential to more people around the world, they need to become reliable. This outage is a serious warning for future dependence on those connected services. If the media can't even report it, how can we expect anyone to do anything to fix or mitigate it? And that, on top of eBay's latest... apparently they've provided sellers some new way to block bids from buyers who haven't registered a validated PayPal account with eBay... but I don't see any way to *register* such an account, and *neither party notified me [a bidder] about the new policy*. Huh? Cheers -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Subscribe/Notify MWI not working for non-numeric accounts w/X-Lite
ok this is a wired problem. when i use X-Lite - after i register with asterisk X-lite sends a subscribe/notify request to asterisk to determine if the account has any messages waiting. if i create a sip.conf account using: user 12345 with a voicemail box 12345 - MWI works user jwolosuk with a voicemail box 12345 MWI fails (gets a 404 not found upon a subscribe) does anyone have a clue why this doesn't work or what i need to set in order for this to work? feedback is appreciated, -- --- John C. Wolosuk Jr. Unix/Linux Systems Administrator Academic Computing Communications Center University of Illinois @ Chicago E-Mail: jwolosuk at uic dot edu --- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] A102 card, BT ISDN30e, silence
On Aug 17, 2007, at 5:55 AM, Rory Campbell-Lange wrote:
Summary:
Can't hear incoming/outgoing calls to/from ISDN over Sangoma A102
card.
Debian stable + Asterisk 1:1.2.13.
Thanks for the response, Andres. We've changed the timing source but
still no joy. This is very odd; calling between internal AIX/SIP
extensions works fine.
I am not sure when the zaptel conf gets read,
make sure to unload - re-load the drivers
If anyone else can think of anything, I would be grateful to hear.
if your card has echo cancel and you are also canceling echo in your
conf, that's a problem
make sure your telco agrees with your channels, like d being in the 16th
try using sangoma tools (wanpipemon) and even calling sangoma,
Rory
On 16/08/07, Andres Paglayan ([EMAIL PROTECTED]) wrote:
On Aug 16, 2007, at 8:58 AM, Rory Campbell-Lange wrote:
However both incoming and outgoing calls are greeted by silence.
...
Zaptel
--
loadzone=uk
defaultzone=uk
#Sangoma A101 port 1 [slot:3 bus:2 span: 1]
span=1,0,0,ccs,hdb3,crc4
the first 0 is the timing source,
if you are connecting this span to a telco
it should be 1 for your telco to provide timing,
which is what they expect
; note that only 25 channels are active
bchan=1-15,17-31
dchan=16
Zapata
--
-
[channels]
language=en
context=from-pstn
signalling=pri_cpe
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=no
echotraining=800
rxgain=0.0
txgain=0.0
group=0
callgroup=1
pickupgroup=1
immediate=no
pridialplan=unknown
prilocaldialplan=unknown
group=1
callerid=asreceived
;Sangoma A101 port 1 [slot:3 bus:2 span: 1]
switchtype=euroisdn
context=from-pstn
group=0
signalling=pri_cpe
channel = 1-15,17-31
Extensions
--
-
[general]
static=yes
writeprotect=no
autofallthrough=yes
clearglobalvars=no
priorityjumping=no
[globals]
CONSOLE=Console/dsp ; Console interface for demo
IAXINFO=guest ; IAXtel username/password
TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0)
[dundi-e164-local]
switch = DUNDi/e164
[local]
ignorepat = 9
include = default
include = parkedcalls
[default]
exten = 001,1,Dial(SIP/001,5)
exten = 001,n,VoiceMail([EMAIL PROTECTED])
exten = 001,n,Playback(vm-goodbye)
exten = 001,n,Hangup
exten = 901,1,Dial(IAX2/901,5)
exten = 901,n,Hangup
exten = 100,1,Dial(SIP/100,5)
exten = 100,n,VoiceMail([EMAIL PROTECTED])
exten = 100,n,Playback(vm-goodbye)
exten = 100,n,Hangup
exten = _0XX,1,Dial(Zap/g0/${EXTEN})
exten = _ZX,1,Dial(Zap/g0/${EXTEN})
[from-pstn]
; BT pass in 6 digits only?
exten = 032685,1,Answer()
exten = 032685,2,Playback(demo-echotest)
exten = 032685,3,Echo()
exten = 032685,4,Playback(demo-echodone)
exten = 032685,5,Hangup()
--
Rory Campbell-Lange
Director
Campbell-Lange Workshop Ltd.
[EMAIL PROTECTED]
www.campbell-lange.net
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Bapak
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Rory Campbell-Lange
Director
Campbell-Lange Workshop Ltd.
[EMAIL PROTECTED]
www.campbell-lange.net
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Re: [asterisk-users] Lock extension from asterisk
On Aug 17, 2007, at 6:52 AM, Doug Lytle wrote:
Gordon Henderson wrote:
On Fri, 17 Aug 2007, Andres Jimenez wrote:
exten = ,1,Answer()
exten = ,n,Set(me=${CALLERID(num)})
exten = ,n,Set(DB(${me}/locked)=1)
exten = ,1,Answer()
exten = ,n,Set(me=${CALLERID(num)})
exten = ,n,VMAuthenticate(${me})
exten = ,n,Set(DB(${me}/locked)=)
This is good, but it can be done with just 1 extension:
exten = 80*,1,Answer()
exten = 80*,2,Set(LOCKED=${DB(phonelocked/${CALLERID(number)})})
exten = 80*,3,GotoIf($[${LOCKED} = YES]?80*,4:80*,101)
exten = 80*,4,Set(DB(phonelocked/${CALLERID(number)})=NO)
exten = 80*,5,Playback(de-activated)
exten = 80*,6,Hangup()
exten = 80*,101,Set(DB(phonelock/${CALLERID(number)})=YES)
exten = 80*,102,Playback(activated)
exten = 80*,103,Hangup()
Guys, very nice dialplan programming,
as a user's opinion, the two extension approach might be better.
so the user doesn't need to remember whether the phone is locked or not,
and accidentally lock it when the contrary was meant,
(unless you send some sip text to the phone display)
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little
Temporary Safety, deserve neither Liberty nor Safety.
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Re: [asterisk-users] Call Limits
At 06:37 AM 8/17/2007, you wrote: Some of my asterisk users have used their maximum call limit for incoming calls (peers). There incoming call limit should automatically reset to zero after hangup but its not happening and they no longer can recieve any calls as their allowed limit is already full. So is there any way to reset the call limit on peers by commands or do i have to restart my asterisk server? It's not just you, it happens to my wife too. No rhyme or reason I can see, I just try to restart asterisk occasionally so it doesn't get that far. Ira ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Limits
I think its an Asterisk bug, call-limits stopped working for me once I upgraded from 1.2.16 to 1.2.18. There is a bug opened for it, but the issue hasn't been resolved yet. Here is the link: http://bugs.digium.com/view.php?id=9794 -Remi Ira wrote: At 06:37 AM 8/17/2007, you wrote: Some of my asterisk users have used their maximum call limit for incoming calls (peers). There incoming call limit should automatically reset to zero after hangup but its not happening and they no longer can recieve any calls as their allowed limit is already full. So is there any way to reset the call limit on peers by commands or do i have to restart my asterisk server? It's not just you, it happens to my wife too. No rhyme or reason I can see, I just try to restart asterisk occasionally so it doesn't get that far. Ira ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Patent issues, what features we can't use?
Jay R. Ashworth wrote: On Thu, Aug 16, 2007 at 11:00:05PM -0400, Zeeshan Zakaria wrote: This is really ridiculous. So this means that now nobody can use fax-to-email without paying to J2 first? Horseshit. Prior art is trivial. How old is Hylafax? Cheers, -- jra It's never trivial if you're a small company. J2 has already won settlements from several smaller companies, which gives it precedence. Once precedence is established, it's almost a done deal for future lawsuits and fighting them is exponentially harder with each settlement they get. While it may boil down in the end to prior art, having the money to fight that far in the legal system is something else. A fight like that would put most small businesses under, and forget about getting external funding if you have this hanging over your head. No one wants to fund a company with a lawsuit against it. N. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Limits
It's a bug and you should probably upgrade, but yes, restarting will resolve the problem temporarily. On 8/17/07, Ira [EMAIL PROTECTED] wrote: At 06:37 AM 8/17/2007, you wrote: Some of my asterisk users have used their maximum call limit for incoming calls (peers). There incoming call limit should automatically reset to zero after hangup but its not happening and they no longer can recieve any calls as their allowed limit is already full. So is there any way to reset the call limit on peers by commands or do i have to restart my asterisk server? It's not just you, it happens to my wife too. No rhyme or reason I can see, I just try to restart asterisk occasionally so it doesn't get that far. Ira ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Anthony Cennami ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Patent issues, what features we can't use?
Horseshit. Prior art is trivial. How old is Hylafax? Cheers, -- jra It's never trivial if you're a small company. J2 has already won settlements from several smaller companies, which gives it precedence. Once precedence is established, it's almost a done deal for future lawsuits and fighting them is exponentially harder with each settlement they get. While it may boil down in the end to prior art, having the money to fight that far in the legal system is something else. A fight like that would put most small businesses under, and forget about getting external funding if you have this hanging over your head. No one wants to fund a company with a lawsuit against it. N. The really good news here is that the recent KSR vs. Teleflex ruling as decided by SCOTUS gives you a fair bit of firepower in at least getting this patent reexamined, even with the precedent. I think that J2 would be quite unlikely to try and push a case forward in light of this. Regards, - Brad ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Paging: Does anyone have a simple howto for Polycoms?
At 08:19 8/17/2007, Dave Fullerton wrote:
Doug wrote:
I've looked at the following pages, and they are
just so garbled. I keep going around in circles:
http://www.voip-info.org/wiki/view/Polycom+auto-answer+config
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+page
http://www.voip-info.org/wiki/index.php?page=Asterisk+Paging+and+Intercom
http://lists.digium.com/pipermail/asterisk-users/2006-May/152764.html
http://threebit.net/mail-archive/asterisk-users/msg23241.html
http://www.aussievoip.com.au/wiki/freePBX-Paging
http://www.voip-info.org/wiki/view/Setting+up+paging+with+a+sound+card
Can anyone just show some simple working examples on
1. The Asterisk side
2. The Polycom side
Here's what I use on my production system with 1.2.24. This for one of
our four page zones but it happens to page through the polycoms in the
office.
In extensions.conf I have the following:
[pagezones]
; Office page zone through phones
; I don't want to see page calls in my cdr reports.
exten = _631,1,SetAMAFlags(omit)
; There are actually several more phones in here but I cut them
; out for readability
exten = _631,n,Page(Local/[EMAIL PROTECTED]Local/[EMAIL PROTECTED])
exten = _631,n,Hangup
[intercom]
exten = _133XX,1,Macro(pageextension,SIP/${EXTEN:1})
[macro-pageextension]
; Paging macro:
; Check to see if device is in use and DO NOT PAGE if they are
; ${ARG1} - Device to page
;
exten = s,1,ChanIsAvail(${ARG1}|js) ; j for jump and s is for ANY call
exten = s,n,Set(_ALERT_INFO=page) ; This is for the PolyComs
exten = s,n,Dial(${ARG1}||)
exten = s,n,Hangup
exten = s,102,Hangup
The [pagezones] context is included in each phones context to make it
available. What happens is the page zone extension is dialed, the page
app is called with several local channels in the [intercom] context. All
of these channels will be dumped into a meetme conference where everyone
except the person paging is muted. Since it uses meetme you will need a
zaptel timing device or ztdummy loaded. The line in the intercom context
simply calls a macro (which I borrowed from voip-info.org I believe).
The macro first checks to see if the phone is in use. If it is not, then
the _ALERT_INFO header is set to page (more on this below) and the
phone is then dialed. If the phone is in use then that local channel is
hungup and will not be paged to. Paging a phone that is in use causes
some odd things to happen on both the phone and asterisk side sometimes.
On the polycom side, here's what I have set in the sip.cfg (I'm using 1.6.7)
You must fill in values in for the alertInfo tag. It's near the top of
the config file in the voIpProtSIP section. See section 4.6.1.1.3.2
(page 74) of the SIP 1.6 Admin Guide for details. Here is how I filled
mine out:
alertInfo voIpProt.SIP.alertInfo.1.value=page
voIpProt.SIP.alertInfo.1.class=4/
Notice the alertInfo.1.value is set to page, the same as what I set
_ALERT_INFO to in my macro. The class is set to the ring type I want to
use on the phone. RingType is discussed in section 4.6.1.7.2 (page 91).
Mine is set to 4 which corresponds to:
RING_ANSWER se.rt.4.name=Ring Answer se.rt.4.type=ring-answer
se.rt.4.timeout=500 se.rt.4.ringer=13 se.rt.4.callWait=6
se.rt.4.mod=1/
When one of my phones is paged it rings for 1/2 a second and then
automatically answers the incoming call. I have set se.rt.4.ringer=13
because I have created a custom page beep ring tone. You can use one of
the predefined ring tones or if you don't want any page beep set the
ring class in the alertInfo tag to 3 which is auto-answer.
Hope that answers your question.
-Dave
Thanks, Dave. This looks a bit more clear
than what's up on the wiki.
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Re: [asterisk-users] Call Limits
I have the sample problem. Just turned it off for now. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Remi Quezada Sent: Friday, August 17, 2007 2:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call Limits I think its an Asterisk bug, call-limits stopped working for me once I upgraded from 1.2.16 to 1.2.18. There is a bug opened for it, but the issue hasn't been resolved yet. Here is the link: http://bugs.digium.com/view.php?id=9794 -Remi Ira wrote: At 06:37 AM 8/17/2007, you wrote: Some of my asterisk users have used their maximum call limit for incoming calls (peers). There incoming call limit should automatically reset to zero after hangup but its not happening and they no longer can recieve any calls as their allowed limit is already full. So is there any way to reset the call limit on peers by commands or do i have to restart my asterisk server? It's not just you, it happens to my wife too. No rhyme or reason I can see, I just try to restart asterisk occasionally so it doesn't get that far. Ira ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] gsm errors
Hi iam using Asteriks 1.2.17 Server Side ( provider Side g729) clients side gsm when iam calling, iam getting lot of errors like below and lot of voice breaks Aug 16 21:23:14 WARNING[9521] dsp.c: Inband DTMF is not supported on codec gsm. Use RFC2833 any suggestions ram ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Lock extension from asterisk
On Fri, 17 Aug 2007, Andres Paglayan wrote: Guys, very nice dialplan programming, as a user's opinion, the two extension approach might be better. so the user doesn't need to remember whether the phone is locked or not, and accidentally lock it when the contrary was meant, (unless you send some sip text to the phone display) I've experimented with sending text to a phone display... it's a bit hit miss... and of-course not all phone have textual displays, and some require you to hit a key to display the text. I experimented with it in speed-dial programming - trying to echo te number back to the display. Worked a treat on GXP2000's (untile you got more than one message), but on the Snom 300 it just seemed fiddly to get the message on the display. Gordon ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Lock extension from asterisk
2007/8/17, Andres Paglayan [EMAIL PROTECTED]:
Guys, very nice dialplan programming,
as a user's opinion, the two extension approach might be better.
so the user doesn't need to remember whether the phone is locked or not,
and accidentally lock it when the contrary was meant,
(unless you send some sip text to the phone display)
In the latest version (see below) I added some playback that will say
if the phone is lock or unlock, before and after locking/unlocking it.
;Locking system
;LOCK
exten = ,1,Answer()
exten = ,n,Set(me=${CALLERID(num)})
exten = ,n,GotoIf(${DB(${me}/locked)}?,101:,201)
exten = ,101,Playback(security)
exten = ,n,Playback(activated)
exten = ,n,VMAuthenticate(${me})
exten = ,n,Set(DB(${me}/locked)=)
exten = ,n,Playback(security)
exten = ,n,Playback(now)
exten = ,n,Playback(de-activated)
exten = ,n,Hangup()
exten = ,201,Playback(security)
exten = ,n,Playback(de-activated)
exten = ,n,VMAuthenticate(${me})
exten = ,n,Set(DB(${me}/locked)=1)
exten = ,n,Playback(security)
exten = ,n,Playback(now)
exten = ,n,Playback(activated)
exten = ,n,Hangup()
--
Andres Jimenez
GPG : http://www.andresin.com/gpg/[EMAIL PROTECTED]
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Re: [asterisk-users] Lock extension from asterisk
Andres Jimenez wrote: In the latest version (see below) I added some playback that will say if the phone is lock or unlock, before and after locking/unlocking it. Just a note, You will want to make sure that (911/999) calls are handled properly when the phone is locked down. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] gsm errors
On Sat, 18 Aug 2007, ram wrote: Hi iam using Asteriks 1.2.17 Server Side ( provider Side g729) clients side gsm when iam calling, iam getting lot of errors like below and lot of voice breaks Aug 16 21:23:14 WARNING[9521] dsp.c: Inband DTMF is not supported on codec gsm. Use RFC2833 any suggestions This might sound obvious, but Use rfc2833. But apart from that, you're transcoding from GSM (a lossy format) to g729 (another lossy format) so audio quality is going to be quite poor. Can't you stick to either GSM or g729 all the way? What clients are you using that only support GSM? (and if they're on a LAN why not use G711/uLaw/aLaw?) Gordon ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] gsm errors
On 8/18/07, Gordon Henderson [EMAIL PROTECTED] wrote: On Sat, 18 Aug 2007, ram wrote: Hi iam using Asteriks 1.2.17 Server Side ( provider Side g729) clients side gsm when iam calling, iam getting lot of errors like below and lot of voice breaks Aug 16 21:23:14 WARNING[9521] dsp.c: Inband DTMF is not supported on codec gsm. Use RFC2833 any suggestions This might sound obvious, but Use rfc2833. But apart from that, you're transcoding from GSM (a lossy format) to g729 (another lossy format) so audio quality is going to be quite poor. Can't you stick to either GSM or g729 all the way? What clients are you using that only support GSM? (and if they're on a LAN why not use G711/uLaw/aLaw?) My provider support only g729 and my client have callcenter Suite which support only GSM any suggestion to come over this problem ram ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] No audio on ISDN PRI calls
Hello, I have a Sangoma A101 connected to an ISDN30 (E1 in the UK) with some Snom 300 and Idefisk softphones. I can do SIP and IAX2 calls just fine, however I cant get any audio in either direction on the Zap channels. When I call in or dial out over the ISDN30 (UK E1) it rings the other end but when answered ,silence follows. I've been provisioned 25 out of the 31 channels only at the moment and I'm using a dummy number given by BT for testing. The is no alarms on the line, nor errors on the CLI or in the /va/log/messages or wanrouter. How could I troubleshoot an audio problem with the Sangoma card? Any ideas? Thank you much. Veselin ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Detecting DTMF Tones from Muted app_meetme Participants
On 8/17/07, David Roden [EMAIL PROTECTED] wrote: Hi, folks. I have a problem using Asterisk 1.2. I create conferences using app_meetme and Zap channels, and for every participant I run the script defined by AGI_BACKGROUND_SCRIPT to be able to listen and react to DTMF tones. As the docs tell me, when using the AGI background script one loses the ability to control the meetme conference via the command line so for muting conference participants I had to create a small workaround (letting the user leave the conference by letting his AGI background script finish) and then rejoin him using the 'm' flag so that he can keep listening but is not allowed to talk. Now, while the user is muted, I can not read any DTMF tones in the AGI background script. Okay, you might say, the user is muted. It makes sense that his DTMF tones do not come through. But I say, the muting, i.e. the nullification of any incoming audio is done somewhere within Asterisk, why is the DTMF tone detection not done _before_ the incoming audio is thrown away? So, basically, my question is, how can I have a muted participant in a meetme conference that can still control his listening experience via DTMF tones? And I'm not afraid to mess around in the code - somehow, somewhere (at least in my imagination) there have to be two lines, the one reading remove_audio(); and the other one reading detect_dtmf_tones(); which should be rearranged... :) Oh, if only it were that simple. There are many places within the app_meetme.c code that parse through DTMF information, and it depends on the type of channel as to whether it is parsed at all, or how it is passed along. I've done quite a bit of messing around with meetme, and most recently I put up a patch that would do better DTMF passthru between VOIP and Zap channels in meetme sessions within Asterisk 1.2.X (a backport of the feature in Asterisk 1.4). http://www.eflo.net/files/meetme_DTMF_passthru-1.2.23.patch Of course I must mention that the Asterisk 1.2 branch is in security-patch-only mode and it is encouraged that you use Asterisk 1.4 to help fix the bugs that are in it and experience some of the feature enhancements that it has. MATT--- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Lock extension from asterisk
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 2007/8/17, Doug Lytle : Just a note, You will want to make sure that (911/999) calls are handled properly when the phone is locked down. Good point. - -- Andres Jimenez GPG : http://www.andresin.com/gpg/[EMAIL PROTECTED] -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.6 (GNU/Linux) Comment: http://firegpg.tuxfamily.org iD8DBQFGxgMD8SZxpGYWwpYRAvXHAJ9ZsPF6wzEaEn6y/VDfgxvuJdmXkgCfYxrz ZEEXnAqeELULlSqJxqmdaJw= =NvEA -END PGP SIGNATURE- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM400P FXO click sounds
Hi all and thanks for every suggest about my problem, I found that my TDM400P was sharing IRQ with onboard sound device using cat /proc/interrupts, lspci -v and lspci -vb. When I disable all unnecessary hardware on my machine and test it, clicking sounds continue on the line with the same intensity; again using lspci -vb i found that: 01:00.0 VGA compatible controller: VIA Technologies, Inc. Unknown device 3230 (rev 11) (prog-if 00 [VGA]) Subsystem: Micro-Star International Co., Ltd. Unknown device 7253 Flags: bus master, 66MHz, medium devsel, latency 64, IRQ 11 Memory at c000 (32-bit, prefetchable) Memory at dd00 (32-bit, non-prefetchable) Capabilities: [60] Power Management version 2 Capabilities: [70] AGP version 3.0 04:04.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface Subsystem: Unknown device b119:0003 Flags: bus master, medium devsel, latency 64, IRQ 11 I/O ports at be00 Memory at dfaff000 (32-bit, non-prefetchable) Capabilities: [40] Power Management version 2 Now TDM card share IRQ 11 with onboard vga controller. I have a sata raid 1 level running on the box too and cat /proc/interrupts show me: 0: 23572057 0IO-APIC-edge timer 1:196 0IO-APIC-edge i8042 6: 3 0IO-APIC-edge floppy 7: 0 0IO-APIC-edge parport0 8: 0 0IO-APIC-edge rtc 9: 0 0 IO-APIC-level acpi 14: 66 0IO-APIC-edge ide0 209:3663990 0 IO-APIC-level eth0 217: 403070 0 IO-APIC-level libata 225: 95602389 0 IO-APIC-level wctdm NMI: 3824180 LOC: 23572106 23572083 ERR: 0 Disabling ACPI from kernel at boot, the systen cant detect TDM card. Again, running debian etch x86_64. Click sounds continue again. Running zttest I get: --- Results after 55 passes --- Best: 100.00 -- Worst: 67.395020 -- Average: 98.757324 I change from one slot to other and the problem continue. What follow to solve this issue? Thanks for any suggest. Gustavo Gonzalez ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [asterisk-biz] Skype Outage Leaves Millions Speechless
On Fri, 2007-08-17 at 18:22 +0200, Trixter aka Bret McDanel wrote: On 8/17/07, Aleks Clark [EMAIL PROTECTED] wrote: Actually, the crazy p2p connections actually reinforce their algorithm story. If their p2p algorithms have flaked out, it could cause all sorts of trouble. OTOH, I don't think they'd run logins over p2p given the press from days past about how someone cracked the algorithm and could write their own client that ebay cant control, almost makes you wonder if it was an auto-update gone awry to try to change the algorithm. I dont know what the default setting is, but do know that skype can be set to auto-update itself, which means that some may have been affected while others werent for that reason alone. I am certain though that skype wouldnt admit if it was this, and its likely that any front line people at skype wouldnt know one way or the other for sure what is broke. Imagine if the world's largest online marketplace operated the world's largest alternative (and one of the largest in general) telco and an unregulated global online banking monopoly. And the telco suddenly went down, unexplained, for hours or days. That sounds like a serious threat to global economy and security, right? eBay is that marketplace, owns Skype, that telco, owns PayPal, that bank. This outage should be screaming from the headlines. As those three essential services become essential to more people around the world, they need to become reliable. This outage is a serious warning for future dependence on those connected services. If the media can't even report it, how can we expect anyone to do anything to fix or mitigate it? -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Quick DUNDi Poll Questions, For All Asterisk Users, Please Give Feedback
Questions: 1. Is the wiki DUNDi example and the dundi.conf file too difficult to follow for new users? 2. Does the complexity of the DUNDi setup discourage you from using it or even attempting to configure it? 3. If there was a simple tutorial, step by step guide with easy to setup and test examples, would this encourage more users to investigate and use DUNDi? I'm interested in putting together a new-user tutorial about DUNDi configuration and setup. There is a lot of great information, setup guides already but the feedback I get is that the current examples are a bit complicated to follow for new users. Your feedback is appreciated. Thanks. JR -- JR Richardson Engineering for the Masses ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Quick DUNDi Poll Questions, For All Asterisk Users, Please Give Feedback
1. Yes 2. Yes 3. Yes Nice sales pitch, sounds like one of those late night get rich now! schemes. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of JR Richardson Sent: Friday, August 17, 2007 4:35 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Quick DUNDi Poll Questions, For All Asterisk Users, Please Give Feedback Questions: 1. Is the wiki DUNDi example and the dundi.conf file too difficult to follow for new users? 2. Does the complexity of the DUNDi setup discourage you from using it or even attempting to configure it? 3. If there was a simple tutorial, step by step guide with easy to setup and test examples, would this encourage more users to investigate and use DUNDi? I'm interested in putting together a new-user tutorial about DUNDi configuration and setup. There is a lot of great information, setup guides already but the feedback I get is that the current examples are a bit complicated to follow for new users. Your feedback is appreciated. Thanks. JR -- JR Richardson Engineering for the Masses ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Quick DUNDi Poll Questions, For All Asterisk Users, Please Give Feedback
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of JR Richardson Sent: Friday, August 17, 2007 4:35 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Quick DUNDi Poll Questions, For All Asterisk Users,Please Give Feedback Questions: 1. Is the wiki DUNDi example and the dundi.conf file too difficult to follow for new users? 2. Does the complexity of the DUNDi setup discourage you from using it or even attempting to configure it? 3. If there was a simple tutorial, step by step guide with easy to setup and test examples, would this encourage more users to investigate and use DUNDi? I'm interested in putting together a new-user tutorial about DUNDi configuration and setup. There is a lot of great information, setup guides already but the feedback I get is that the current examples are a bit complicated to follow for new users. Your feedback is appreciated. Thanks. JR -- JR Richardson Engineering for the Masses JR, I'd love to see some tutorials on how to setup DUNDi that are aimed at people who have little experience with DUNDi (that's me). Bobby ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Limits
Upgrade to what? I'm running the most recent 1.2 version and I can't run 1.4 because every 3 calls I have to reboot the machine. Ira At 11:00 AM 8/17/2007, you wrote: It's a bug and you should probably upgrade, but yes, restarting will resolve the problem temporarily. On 8/17/07, Ira mailto:[EMAIL PROTECTED] [EMAIL PROTECTED] wrote: At 06:37 AM 8/17/2007, you wrote: Some of my asterisk users have used their maximum call limit for incoming calls (peers). There incoming call limit should automatically reset to zero after hangup but its not happening and they no longer can recieve any calls as their allowed limit is already full. So is there any way to reset the call limit on peers by commands or do i have to restart my asterisk server? It's not just you, it happens to my wife too. No rhyme or reason I can see, I just try to restart asterisk occasionally so it doesn't get that far. Ira ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Quick DUNDi Poll Questions, For All Asterisk Users, Please Give Feedback
JR Richardson wrote: Questions: 1. Is the wiki DUNDi example and the dundi.conf file too difficult to follow for new users? 2. Does the complexity of the DUNDi setup discourage you from using it or even attempting to configure it? 3. If there was a simple tutorial, step by step guide with easy to setup and test examples, would this encourage more users to investigate and use DUNDi? I'm interested in putting together a new-user tutorial about DUNDi configuration and setup. There is a lot of great information, setup guides already but the feedback I get is that the current examples are a bit complicated to follow for new users. Your feedback is appreciated. Thanks. JR JR I just got DUNDi up and going it the past couple of weeks. Your whitepaper on DUNDi and realtime clustering was the only way I would have gotten it up. Everything else on the wiki is simply too complicated and not well explained. And it still wasn't a piece of cake, even with you document. So yes, additional cookbook type documents that thoroughly explain things should greatly help adoption. Mike Clark ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Patent issues, what features we can't use?
What exactly is patented by J2? Is it receiving fax over the Internet, converting to PDF and sending as an email attachment using sendmail or postfix etc? Or is it receiving it through PRI, or PSTN line over the computer and converting and emailing? What exactly they don't want others to do? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Zaptel 1.2.20 and 1.4.5 released
The Asterisk.org development team has announced the release of Zaptel versions 1.2.20 and 1.4.5. These releases are maintenance releases that fix various known issues. See the ChangeLog included in the releases for a full list of changes. The ChangeLogs are also available separately on the ftp site. This release also contains support for Digium's new 32 channel hardware echo canceler (VPMADT032) for the TDM800P and TDM2400P. Warning to TDM800P and TDM2400P users with FXO modules: Unless your TDM card contains a VPM100M echo canceler, you will notice an increase in your volume levels after upgrading to this release of Zaptel. You may wish to compensate for this change in zapata.conf. Both releases are available as a tarball as well as a patch against the previous release. They are available for download from downloads.digium.com. Thank you for your support! ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RAW asterisk!
Bill Andersen wrote: I'm a network admin that maintains 3 commercial Asterisk servers for my employer. I am wanting to move away from the pre-packaged commercial PBXs to a more pure asterisk setup. The systems I have utilize a nice web GUI to make changes, but it really limits what I can do beyond what they have programmed into their GUI. Bill, If you like working from Windows, you can also check out DialplanPro. I've been using it for our few (so far) clients and our personal phone system. http://www.datatrakpos.com/pos/datatalk/Default.aspx I wrote it to be more of a swiss army knife for Asterisk. I like to use the GUI widgets and visual menu builder to build the basic dialplan menus then use the editor (basic syntax highlighting, parameter suggestions, etc) to write custom scripts using either traditional flat asterisk script or AEL2 and INCLUDE them in the final project scripts which can be automatically uploaded to the server. I also use it to parse my AEL2 scripts remotely from my windows computer using a hook into the aelparse executable written by murph. Its still beta, but mostly because it doesn't yet have all the features I want to eventually include in it. Also, its commercial software or will be someday. -- Warm Regards, Lee ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Channel as MusicOnHold
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, I have what sounds like a simple requirement to add a feature to an existing dial plan context. Currently the user gets a normal .wav file played to them via MusicOnHold while they go through a basic menu prompt. However I now need to take the background audio for this menu from the audio stream of a seperate connected call in another context. Now I've looked at using ChanSpy() but I can't see an option to 'spy as background audio'. I've also looked at the MusicOnHold streaming via a unix script / pipe but that doesn't really help either as I need to somehow copy the other channel audio into the stdin of the pipe. Anyone got any ideas as I'm stuck on this one! Vince. -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (GNU/Linux) Comment: Using GnuPG with Fedora - http://enigmail.mozdev.org iD8DBQFGxkYFPx/nyuA99rgRAo/XAJsGTzONVDrcyv+qpEP3llx7lQ6LkQCgpJ+4 tjURS1gA5yr7xY5qJOr/4VE= =015x -END PGP SIGNATURE- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RAW asterisk!
At 19:35 8/17/2007, Lee Jenkins wrote: Bill Andersen wrote: I'm a network admin that maintains 3 commercial Asterisk servers for my employer. I am wanting to move away from the pre-packaged commercial PBXs to a more pure asterisk setup. The systems I have utilize a nice web GUI to make changes, but it really limits what I can do beyond what they have programmed into their GUI. Bill, If you like working from Windows, you can also check out DialplanPro. I've been using it for our few (so far) clients and our personal phone system. http://www.datatrakpos.com/pos/datatalk/Default.aspx I wrote it to be more of a swiss army knife for Asterisk. I like to use the GUI widgets and visual menu builder to build the basic dialplan menus then use the editor (basic syntax highlighting, parameter suggestions, etc) to write custom scripts using either traditional flat asterisk script or AEL2 and INCLUDE them in the final project scripts which can be automatically uploaded to the server. I also use it to parse my AEL2 scripts remotely from my windows computer using a hook into the aelparse executable written by murph. Its still beta, but mostly because it doesn't yet have all the features I want to eventually include in it. Also, its commercial software or will be someday. -- Warm Regards, Lee Keeewwwl...Delphi! However, all I can get it to do is generate errors: == Application... Start Date : 08/17/2007 20:20:27 Name/Description: astclient.exe Version Number : 0.9.6.75 Exception... Date : 08/17/2007 20:22:40 Address: 00409A5A Module : astclient.exe Type : EConvertError Message: '' is not a valid integer value. Active Controls... Form Class : TfrmOutput Form Text: Dialplan Output Control Class: TCheckBox Control Text : Verbose Commenting Computer... Name: Total Memory: 990 Mb Free Memory : 318 Mb Total Disk : 5.85 Gb Free Disk : 5.67 Gb Operating System... Type: Microsoft Windows 2000 Build # : 2195 Language: English (United States) == ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Quick DUNDi Poll Questions, For All Asterisk Users, Please Give Feedback
On Friday 17 August 2007 04:34:33 pm JR Richardson wrote: Questions: 1. Is the wiki DUNDi example and the dundi.conf file too difficult to follow for new users? it's a bit complicated, though it seems to make sense for large-scale ops. 2. Does the complexity of the DUNDi setup discourage you from using it or even attempting to configure it? what would also encourage me is if dundi.com would be updated, etc. when setting up dundi, i feel as if i'm entering an underground secret society where no one really wants to share much. this leads me to feel as though it's not something that's actively developed/used in the usa, outside of (intra)corporations. i see lots of e164 dundi activity in europe, but not here, why? 3. If there was a simple tutorial, step by step guide with easy to setup and test examples, would this encourage more users to investigate and use DUNDi? this would help a lot. also, since dundi is peer based, it would be nice if dundi.com or some central place could get new users introduced to finding peers, etc. even people on the dundi.com list of tier 1 peers don't respond to emails, etc. i heard your talk today! thanks again for your help. -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Manager Proxy - Still required?
Hi, With more recent version of v1.2 and with v1.4 are things like the AstManProxy still recommended if you want to have a bunch of applications talking directly to Asterisk? Cheers! -- Andrew Ruthven, Wellington, New Zealand At work: [EMAIL PROTECTED] At home: [EMAIL PROTECTED] GPG fpr: 34CA 12A3 C6F8 B156 72C2 D0D7 D286 CE0C 0C62 B791 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel 1.2.20 and 1.4.5 released
On Fri, Aug 17, 2007 at 06:15:28PM -0500, Asterisk Development Team wrote: The Asterisk.org development team has announced the release of Zaptel versions 1.2.20 and 1.4.5. These releases are maintenance releases that fix various known issues. See the ChangeLog included in the releases for a full list of changes. The ChangeLogs are also available separately on the ftp site. Old habits die hard :-) Anyway, that version also includes some changes intended to help cross-builders and packages. Please report bugs... -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Quick DUNDi Poll Questions, For All Asterisk Users, Please Give Feedback
On Fri, Aug 17, 2007 at 04:34:33PM -0500, JR Richardson wrote: Questions: 1. Is the wiki DUNDi example and the dundi.conf file too difficult to follow for new users? You imply that both need fixing 2. Does the complexity of the DUNDi setup discourage you from using it or even attempting to configure it? 3. If there was a simple tutorial, step by step guide with easy to setup and test examples, would this encourage more users to investigate and use DUNDi? And then suggest to fix the problem elsewhere. If there was such a tutorial, it would be half-frustrated before I would have even heard of it. Which brings another set of questions: 1. Is it difficult to add documentation to the wiki? 2. Is it difficult to add documentation to the Asterisk source? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Patent issues, what features we can't use?
Zeeshan Zakaria wrote: What exactly is patented by J2? Is it receiving fax over the Internet, converting to PDF and sending as an email attachment using sendmail or postfix etc? Or is it receiving it through PRI, or PSTN line over the computer and converting and emailing? What exactly they don't want others to do? I think you have this backwards. It isn't a matter of what they don't want others to do. Its a matter of what they do want them to do - pay. As to what is patented, they have a number of patents on various aspects of FAX PSTN to packet network integration. Steve ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Channel as MusicOnHold
Use a MeetMe room -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Vincent Sweeney Sent: Friday, August 17, 2007 9:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk Channel as MusicOnHold -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, I have what sounds like a simple requirement to add a feature to an existing dial plan context. Currently the user gets a normal .wav file played to them via MusicOnHold while they go through a basic menu prompt. However I now need to take the background audio for this menu from the audio stream of a seperate connected call in another context. Now I've looked at using ChanSpy() but I can't see an option to 'spy as background audio'. I've also looked at the MusicOnHold streaming via a unix script / pipe but that doesn't really help either as I need to somehow copy the other channel audio into the stdin of the pipe. Anyone got any ideas as I'm stuck on this one! Vince. -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (GNU/Linux) Comment: Using GnuPG with Fedora - http://enigmail.mozdev.org iD8DBQFGxkYFPx/nyuA99rgRAo/XAJsGTzONVDrcyv+qpEP3llx7lQ6LkQCgpJ+4 tjURS1gA5yr7xY5qJOr/4VE= =015x -END PGP SIGNATURE- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
