Re: [asterisk-users] Crash in Asterisk
On Thu, 7 Jan 2010 15:58:43 -0430 Danny Dias ing.diasda...@gmail.com wrote: My friends, I'm having some problems in my Asterisk, the thing is that Asterisk seem to be crashed (or dead) sometimes (2 times in 3 weeks) [Jan 5 16:51:19] WARNING[6787] channel.c: Channel allocation failed: Refusing due to active shutdown Hmm. Seems to start here. Are you sure someone isn't just restarting mysql? I'd find out what causes this string to be issued in channel.c, no? [Jan 5 16:51:54] ERROR[6787] res_config_mysql.c: MySQL RealTime: Ping failed (2003). Trying an explicit reconnect. But that's pretty obviously mysql unavailable. [Jan 5 16:51:54] ERROR[6787] res_config_mysql.c: MySQL RealTime: Failed to connect database server dreampbx on 127.0.0.1 (err 2003). Check debug for more info. mysql on the same machine, that is? So have a look there? [Jan 5 16:51:54] ERROR[6787] res_config_mysql.c: MySQL RealTime: Failed to connect database server dreampbx on 127.0.0.1 (err 2003). Check debug for more info. What do you think my friends? How can i solve this problem? Find out why mysql isn't talking to asterisk, or don't use mysql realtime? Just my guess. Good luck, -- Michael Higgins ___ -- 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] Can't dial out until I dial in once
On Sat, 25 Apr 2009 00:01:44 -0400 Michael van der Stoop mst...@dpia.ca wrote: I call in once from a cell phone, which is successful then I can call out with out issue. It's a bug. Maybe this one? http://bugs.digium.com/print_bug_page.php?bug_id=14577 Cheers, -- |\ /|| | ~ ~ | \/ ||---| `|` ? ||ichael | |iggins\^ / michael.higgins[at]evolone[dot]org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium fax force T38?
On Mon, 27 Apr 2009 00:33:44 +1200 Michael as...@nettrust.co.nz wrote: I can't with Digium fax, and it always fails at the point it decides to switch to T38. Have you tried dedicating the line to fax only, no detection? I tried using it, but for me it apparently fails the codec switch: WARNING[3862]: frame.c:214 __ast_smoother_feed: Smoother was working on 4 format frames, now trying to feed 64? [Apr 14 21:50:46] ERROR[3862]: res_fax.c:910 generic_fax_exec: channel 'DAHDI/1-1' fax session '3' failure, reason: 'Failed to feed the smoother' But if I just initiate the fax on that channel for whatever comes, it works. http://forums.digium.com/viewtopic.php?p=128681#128681 Cheers, -- |\ /|| | ~ ~ | \/ ||---| `|` ? ||ichael | |iggins\^ / michael.higgins[at]evolone[dot]org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Can I tell if a call picked up on PSTN extension... for example?
Don't know enough to properly term the problem I'm seeing... sorry if subject
appears vague. And I have other questions too, but Newbie Help Wanted isn't
exactly more specific... ;-)
My setup, intended for testing and all, * version 1.6.0.6, dahdi with an
X100p clone. Regular phone line provides PSTN access with one port (and my DSL).
Calls come in and are passed to 'linphone' on a laptop inside my local network.
Everything works great with IAX2 and SIP peers, in that, calls end and linphone
reports 'call terminated', or whatever. All *so* very cool! '-)
But, when someone calls on the land line, there are issues. First, I admit to
having virtually no clue what I'm doing... but, here goes:
When the caller hangs up, linphone doesn't know. This is an annoyance, more
than anything, perhaps, as I'm not sure yet if it shows busy on the other end.
And, to the Subject:, when a call comes in, if someone should pick up on a
phone attached on the PSTN line elsewhere, the dialplan continues, announces
prompts and records a voice mail...
Should it, or could it even, detect if the line was picked up elsewhere?
There's only one port on the card.
I have a feeling there are glaring errors in my setup, so hope I'm not asking
too much indulgence from the list to have a look... IOW, I'm keen to test in
the latest branch moving forward, so thinking maybe asking a little help on
this glorified home answering machine setup isn't *too* far out of line. '-)
. . .
Here are my configs, hopefully well sanitized...
# cat /etc/asterisk/chan_dahdi.conf
[trunkgroups]
[channels]
toneduration=250
usecallerid=yes
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=no
echotraining=yes
rxgain=10
txgain=10
group=1
callgroup=1
pickupgroup=1
immediate=yes
ringtimeout=8000
signalling=fxs_ls
callerid=asreceived
channel = 1
# cat /etc/asterisk/extensions.conf
[general]
static=yes
writeprotect=no
clearglobalvars=no
[globals]
CONSOLE=Console/dsp
TRUNK=DAHDI/g1
TRUNKMSD=0
[trunktollfree]
exten = _91800NXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
exten = _91888NXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
exten = _91877NXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
exten = _91866NXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
[macro-stdexten]
exten = s,1,Dial(${ARG2},20,m)
exten = s,2,Goto(s-${DIALSTATUS},1)
exten = s-BUSY,1,Voicemail(${ARG1},b)
exten = s-BUSY,2,Goto(default,s,5)
exten = s-NOANSWER,1,Voicemail(${ARG1},u)
exten = s-NOANSWER,2,Goto(default,s,5)
exten = _s-.,1,Goto(s-NOANSWER,1)
[macro-trunkdial]
exten = s,1,Dial(${ARG1})
exten = s,n,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Hangup
exten = s-BUSY,1,Hangup
exten = _s-.,1,NoOp
[page]
exten = _X.,1,Macro(page,SIP/${EXTEN})
[opensky]
exten = _1.,1,NoOp('opensky dial')
exten = _1.,2,Dial(SIP/${ext...@gizmo5|120|j)
exten = _1.,3,Hangup()
[default]
include = opensky
;# interrupt to voicemail
exten = a,1,VoicemailMain(${mbx})
exten = a,n,Hangup()
;# Answering machine...???
exten = s,1,Verbose(1|dumb answering machine|${CALLERIDNUM})
; pick up local extension?
exten = s,2,Wait(10)
exten = s,3,Answer()
exten = s,4,Macro(stdexten,,SIP/linphone)
exten = s,5,Hangup()
exten = s,6,SoftHangup(SIP/linphone)
exten = s,7(fail),Hangup
exten = 600,1,Playback(demo-echotest)
exten = 600,n,Echo
exten = 600,n,Playback(demo-echodone)
exten = 600,n,Hangup()
;# record message
exten = 1205,1,Wait(2)
exten = 1205,2,Record(/tmp/asterisk-recording:gsm)
exten = 1205,3,Hangup
exten = _X.,1,Dial(DAHDI/1/${EXTEN})
exten = _X.,n,Hangup()
;# opensky gizmo5
exten = _1333.,1,Goto(opensky,,1)
exten = _333.,1,Goto(opensky,,1) ;COPY THIS CONFIG
exten = _skype[_].,1,Goto(opensky,,1) ;INTO YOUR
exten = 563,1,Dial(SIP/skype_echo...@proxy01.sipphone.com) ;extensions.conf
; transfer all to cell phone
exten = ,1,Wait(1)
exten = ,n,Dial(DAHDI/1/*72XX,,)
exten = ,n,Wait(1)
exten = ,n,Hangup()
exten = ,1,Dial(SIP/linphone,20)
exten = ,n,Hangup()
; straight to voicemail
exten = ,1,Voicemail(6...@default)
exten = ,n,Hangup()
exten = 8500,1,VoicemailMain
exten = 8500,n,Hangup
# cat /etc/asterisk/iax.conf
[general]
bandwidth=low
disallow=lpc10
jitterbuffer=no
forcejitterbuffer=no
autokill=yes
[guest]
type=user
context=default
callerid=Guest IAX User
[iaxtel]
type=user
context=default
auth=rsa
inkeys=iaxtel
[iaxfwd]
type=user
context=default
auth=rsa
inkeys=freeworlddialup
[demo]
type=peer
username=asterisk
secret=supersecret
host=216.207.245.47
[]
type=user
insecure=very
context=default
context=default
disallow=all
allow=ulaw
allow=alaw
allow=gsm
# cat
[asterisk-users] Test asterisk from behind my firewall
I have an asterisk server at home. I'd like to test one just installed elsewhere. Both servers are behind firewalls. I can see the session start in CLI, my congratulations is apparently playing and RTP is being sent. Hearing no audio. Can send key presses and see audio playing changed. Peer audio RTP is at port 198.145.28.177:10180, but that never shows at the client side, behind a linksys wrt54g, ver 1. w/ latest firmware update. My belief is this should be possible, as the SIP phone is registered to my asterisk box inside my home network, asterisk should stay in the middle and forward the RTP packets to my laptop... am I totally off base? If so, what are some key elements to make that happen? I'll stop now, before I get ignored for being too verbose. '-) Cheers, -- |\ /|| | ~ ~ | \/ ||---| `|` ? ||ichael | |iggins\^ / michael.higgins[at]evolone[dot]org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Test asterisk from behind my firewall [SOLVED]
On Mon, 16 Mar 2009 23:00:32 -0700 Michael Higgins li...@evolone.org wrote: I have an asterisk server at home. I'd like to test one just installed elsewhere. And did succeed just after emailing, of course. :( Sorry for the noise! -- |\ /|| | ~ ~ | \/ ||---| `|` ? ||ichael | |iggins\^ / michael.higgins[at]evolone[dot]org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] clone X100p+dahdi dial out works only after receiving call
On Tue, 3 Mar 2009 12:04:48 +0200 Tzafrir Cohen tzafrir.co...@xorcom.com wrote: http://bugs.digium.com/view.php?id=14577 ? Totally. Switching to fxsls from fxsks did the trick. Do I know what difference this makes to me? Not at all... but it 'fixes' the problem CHANUNAVAIL when it *should* be good to go. http://bugs.digium.com/view.php?id=13927 : Setting them to fxsls works around it by ignoring the hook state. And here: http://bugs.digium.com/view.php?id=13786 Exactly the same fault as xrobau. Unable to make calls out of the FXO ports until the line has rung once. The fault is 100% reproducible by rebooting the box, and the same cycle starts over again. Have found a workaround which is switching from FXSKS to FXSLS for moment which is showing 100% success. And, at the bottom: Already fixed before 1.6.0-rc3, indeed. asterisk -V Asterisk 1.6.0.3 ... so, it should, or should not, be fixed in this case? If fixed in -rc3, .3 is the actual release of the candidate, no? Or, surely it is applied in 1.6.0.6... I bump the ebuild to the later version, reverse my config changes and see what happens. No change, though my SIP client sees 'call declined', or something, rather than 'service not avail'. Same problem, though, and still worked around by using -ls not -ks. Thanks for the clues toward a workaround... though is there some reason I should prefer 'kewl' over 'loop', or whatever this workaround implies? Cheers, -- |\ /|| | ~ ~ | \/ ||---| `|` ? ||ichael | |iggins\^ / michael.higgins[at]evolone[dot]org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] clone X100p+dahdi dial out works only after receiving call
On Sat, 28 Feb 2009 21:52:46 +0200
Tzafrir Cohen tzafrir.co...@xorcom.com wrote:
On Sat, Feb 28, 2009 at 11:24:53AM -0800, Michael Higgins wrote:
So, tweaking configs, rebuilding this and that... restarting,
twiddling, it works (yeah!), but fails on re-boot to work at all.
Consistently, though.
I believe it comes down to this: I can call out only *after* I've
received a call.
[8]
modprobe wctc4xxp
Why? Do you have a transcoder card?
Doh! No, I suppose I don't.
modprobe wcfxo
[8]
So, by chance, instead of ripping my hair for a bit, just to be
sure it's still working *at all*, I call myself:
starting simple switch on 'DAHDI/1-1'
[Feb 28 11:00:49] NOTICE[2458]: chan_dahdi.c:7125 ss_thread: Got
event 18 (Ring Begin)... == Starting DAHDI/1-1 at from-pstn,s,1
failed so falling back to exten 's' == Starting DAHDI/1-1 at
from-pstn,s,1 still failed so falling back to context 'default'
asterisk -rx 'dialplan show s...@from-pstn'
asterisk -rx 'dialplan show s...@from-pstn'
There is no existence of 'from-pstn' context
Command 'dialplan show s...@from-pstn' failed.
Um... is that a clue? '-)
Can I just stick everything in 'default', or do I need different contexts?
-- Executing [...@default:1] Verbose(DAHDI/1-1, 1|dumb
answering machine) in new stack 1|dumb answering machine
-- Executing [...@default:2] Answer(DAHDI/1-1, ) in new stack
-- Executing [...@default:3] Playback(DAHDI/1-1,
transfer,skip) in new stack -- DAHDI/1-1 Playing
'transfer.gsm' (language 'en') -- Executing [...@default:4]
Dial(DAHDI/1-1, SIP/mykhy...@192.168.0.100,20,rt) in new stack
== Using SIP RTP CoS mark 5 -- Called mykhy...@192.168.0.100
-- SIP/192.168.0.100-0827a188 is ringing
-- SIP/192.168.0.100-0827a188 answered DAHDI/1-1
== Spawn extension (default, s, 4) exited non-zero on 'DAHDI/1-1'
-- Hungup 'DAHDI/1-1'
And I get my call... with success.
Now, I try to call out, originate at CLI again:
*CLI originate DAHDI/1/5034735882 extension linphone
== Starting DAHDI/1-1 at default,linphone,1 failed so falling
back to exten 's' -- Executing [...@default:1] Verbose(DAHDI/1-1,
1|dumb answering machine) in new stack 1|dumb answering machine
-- Executing [...@default:2] Answer(DAHDI/1-1, ) in new stack
-- Executing [...@default:3] Playback(DAHDI/1-1,
transfer,skip) in new stack -- DAHDI/1-1 Playing
'transfer.gsm' (language 'en') *CLI -- Executing [...@default:4]
Dial(DAHDI/1-1, SIP/mykhy...@192.168.0.100,20,rt) in new stack
== Using SIP RTP CoS mark 5 -- Called mykhy...@192.168.0.100
-- SIP/192.168.0.100-0827a700 is ringing
-- SIP/192.168.0.100-0827a700 answered DAHDI/1-1
== Spawn extension (default, s, 4) exited non-zero on 'DAHDI/1-1'
-- Hungup 'DAHDI/1-1'
[8]
If you configure things manually, don't also include dahdi-channels.
If you do include it, it is probably best to include it after you set
all the defaults in the lines below.
Okay. I removed it, and just put in the bits generated in that file, less the
reference to a non-existent context line.
Now, the DAHDI context shows as 'default'. Is that okay?
[8]
asterisk -rx 'dialplan show s...@default'
[ Context 'default' created by 'pbx_config' ]
[ Context 'default' created by 'pbx_config' ]
's' =1. Verbose(1|dumb answering machine) [pbx_config]
2. Answer() [pbx_config]
3. Playback(transfer,skip)[pbx_config]
4. Dial(SIP/mykhy...@192.168.0.100,20,rt) [pbx_config]
5. BackGround(asterisk-recording) [pbx_config]
6. Voicemail(6...@default)[pbx_config]
7. Playback(tt-weasels) [pbx_config]
8. Hangup() [pbx_config]
-= 1 extension (8 priorities) in 1 context. =-
asterisk -rx 'dialplan show 6...@default'
[ Context 'default' created by 'pbx_config' ]
'' = 1. Voicemail(6...@default)[pbx_config]
2. Hangup() [pbx_config]
'_X.' = 1. Dial(DAHDI/1/${EXTEN}) [pbx_config]
-= 2 extensions (3 priorities) in 1 context. =-
... and it becomes quite clear I have NO IDEA what I'm doing. Why is that last
line in the dialplan..??
Does any of this make clear why I can't *originate* a call in CLI before
*receiving* a call?
IOW, if I fix this dialplan thing, should the problem go away? Trying to
eliminate other possible (hardware, driver) issues is a distraction at this
point.
originate DAHDI/1/5551212 extension linphone... there a more specific variant
on this in CLI originate command I could use, or something? Like to specify a
context..??
Obviously I need to read up on something, just
Re: [asterisk-users] dialing timing problem?
On Fri, 27 Feb 2009 16:41:16 -0500
Doug Lytle supp...@drdos.info wrote:
Michael Higgins wrote:
exten = _X.,1,Dial(DAHDI/1,${EXTEN})
This should be _X.,1,Dial(DAHDI/g1/${EXTEN})
Doug
Thanks for the help. Indeed it needs to be:
exten = _X.,1,Dial(DAHDI/1/${EXTEN})
I don't know where the comma came in before ${EXTEN}, but that was the deal
killer, leaving me with dialtone and little else.
Maybe it should have thrown a parsing error? Or could there be a use for that
syntax with a comma?
Also, for clarity, putting /g1/, rather than just /1/ killed it too
CHANUNAVAIL, or something.
Cheers,
--
|\ /|| | ~ ~
| \/ ||---| `|` ?
||ichael | |iggins\^ /
michael.higgins[at]evolone[dot]org
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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[asterisk-users] clone X100p+dahdi dial out works only after receiving call
So, tweaking configs, rebuilding this and that... restarting, twiddling, it works (yeah!), but fails on re-boot to work at all. Consistently, though. I believe it comes down to this: I can call out only *after* I've received a call. So, cold boot. Then: modprobe dahdi modprobe wctc4xxp modprobe wcfxo dahdi: Telephony Interface Registered on major 196 dahdi: Version: 2.1.0.3 dahdi_transcode: Loaded. ACPI: PCI Interrupt :00:06.0[A] - Link [LNKB] - GSI 11 (level, low) - IRQ 11 Found a Wildcard FXO: Wildcard X100P cat /proc/dahdi/1 Span 1: WCFXO/0 Wildcard X100P Board 1 (MASTER) 1 WCFXO/0/0 Looks good so far. I think. Don't really know what the strings represent entirely. # /etc/init.d/dahdi start * Starting DAHDI ... Start asterisk: sudo -u asterisk asterisk -cvvv *CLI dahdi show status Description Alarms IRQbpviol CRC4 Fra Codi Options LBO Wildcard X100P Board 1 OK 0 0 0 CAS Unk YEL 0 db (CSU)/0-133 feet (DSX-1) *CLI dahdi show channel 1 Channel: 1 File Descriptor: 10 Span: 1 Extension: Dialing: no Context: from-pstn Caller ID: Calling TON: 0 Caller ID name: Mailbox: none Destroy: 0 InAlarm: 0 Signalling Type: FXS Kewlstart Radio: 0 Owner: None Real: None Callwait: None Threeway: None Confno: -1 Propagated Conference: -1 Real in conference: 0 DSP: no Busy Detection: no TDD: no Relax DTMF: no Dialing/CallwaitCAS: 0/0 Default law: ulaw Fax Handled: no Pulse phone: no DND: no Echo Cancellation: 1 taps (unless TDM bridged) currently OFF Actual Confinfo: Num/0, Mode/0x Actual Confmute: No Hookstate (FXS only): Onhook So, all is good. First test is to see if I can originate a call from CLI: *CLI originate DAHDI/1/5034735882 extension linphone *CLI [Feb 28 10:59:48] NOTICE[2401]: channel.c:3316 __ast_request_and_dial: Unable to request channel DAHDI/1/5034735882 So, by chance, instead of ripping my hair for a bit, just to be sure it's still working *at all*, I call myself: starting simple switch on 'DAHDI/1-1' [Feb 28 11:00:49] NOTICE[2458]: chan_dahdi.c:7125 ss_thread: Got event 18 (Ring Begin)... == Starting DAHDI/1-1 at from-pstn,s,1 failed so falling back to exten 's' == Starting DAHDI/1-1 at from-pstn,s,1 still failed so falling back to context 'default' -- Executing [...@default:1] Verbose(DAHDI/1-1, 1|dumb answering machine) in new stack 1|dumb answering machine -- Executing [...@default:2] Answer(DAHDI/1-1, ) in new stack -- Executing [...@default:3] Playback(DAHDI/1-1, transfer,skip) in new stack -- DAHDI/1-1 Playing 'transfer.gsm' (language 'en') -- Executing [...@default:4] Dial(DAHDI/1-1, SIP/mykhy...@192.168.0.100,20,rt) in new stack == Using SIP RTP CoS mark 5 -- Called mykhy...@192.168.0.100 -- SIP/192.168.0.100-0827a188 is ringing -- SIP/192.168.0.100-0827a188 answered DAHDI/1-1 == Spawn extension (default, s, 4) exited non-zero on 'DAHDI/1-1' -- Hungup 'DAHDI/1-1' And I get my call... with success. Now, I try to call out, originate at CLI again: *CLI originate DAHDI/1/5034735882 extension linphone == Starting DAHDI/1-1 at default,linphone,1 failed so falling back to exten 's' -- Executing [...@default:1] Verbose(DAHDI/1-1, 1|dumb answering machine) in new stack 1|dumb answering machine -- Executing [...@default:2] Answer(DAHDI/1-1, ) in new stack -- Executing [...@default:3] Playback(DAHDI/1-1, transfer,skip) in new stack -- DAHDI/1-1 Playing 'transfer.gsm' (language 'en') *CLI -- Executing [...@default:4] Dial(DAHDI/1-1, SIP/mykhy...@192.168.0.100,20,rt) in new stack == Using SIP RTP CoS mark 5 -- Called mykhy...@192.168.0.100 -- SIP/192.168.0.100-0827a700 is ringing -- SIP/192.168.0.100-0827a700 answered DAHDI/1-1 == Spawn extension (default, s, 4) exited non-zero on 'DAHDI/1-1' -- Hungup 'DAHDI/1-1' So, obviously it works... sort of. I'm assuming that, since I don't really *know* what I'm doing, someone else who *does* can probably point out the missing or incorrect part of my configuration. (Meanwhile, I'll see about IRQ status... seems a possible culprit, now that I think about it more.) Anyway, here's the bits of: ./chan_dahdi.conf ./dahdi-channels.conf ./extensions.conf [trunkgroups] [channels] #include /etc/asterisk/dahdi-channels.conf signalling=fxs_ks usecallerid=yes callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=no echocancelwhenbridged=no group=1 callgroup=1 pickupgroup=1 immediate=yes ringtimeout=8000 signalling=fxs_ks callerid=asreceived group=0 context=from-pstn channel = 1 [general] static=yes writeprotect=no clearglobalvars=no [globals] CONSOLE=Console/dsp IAXINFO=guest TRUNK=DAHDI/g1
[asterisk-users] dialing timing problem?
Preparing to use * for a 'real' installation shortly.
Meanwhile, I've got a single port clone thing, 00:06.0 Communication
controller: Motorola Wildcard X100P working to answer my landline and send
calls to my laptop or voicemail. Sweet!
Trying to call out from linphone, I set up this:
exten = _X.,1,Dial(DAHDI/1,${EXTEN})
Both SIP client and this extension are in default context.
Now, I get a dial tone, but the dialing is just not happening. Some tones are
generated, but not in time. Seems random... Clock issues?
OTOH, I can originate from the CLI to my SIP extension. What does this mean?
Any help with troubleshooting this appreciated. Some info:
/etc/asterisk/sip.conf
context=default
allowoverlap=no
bindport=5060
bindaddr=0.0.0.0
tcpenable=no
tcpbindaddr=0.0.0.0
srvlookup=yes
[authentication]
[basic-options](!)
dtmfmode=rfc2833
context=from-office
type=friend
[natted-phone](!,basic-options)
nat=yes
canreinvite=no
host=dynamic
[public-phone](!,basic-options)
nat=no
canreinvite=yes
[my-codecs](!)
disallow=all
allow=ilbc
allow=g729
allow=gsm
allow=g723
allow=ulaw
[ulaw-phone](!)
disallow=all
allow=ulaw
[linphone]
type=friend
context=default
secret=secret
host=dynamic
dtmfmode=rfc2833
defaultuser=mykhyggz
defaultip=192.168.0.100
/etc/asterisk/chan_dahdi.conf
[trunkgroups]
[channels]
usecallerid=yes
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
signalling=fxs_ks
group=1
channel = 1
#include /etc/asterisk/dahdi-channels.conf
; Autogenerated by /usr/sbin/dahdi_genconf on Thu Feb 26 16:17:56 2009 -- do
not hand edit
; Dahdi Channels Configurations (chan_dahdi.conf)
;
; This is not intended to be a complete chan_dahdi.conf. Rather, it is intended
; to be #include-d by /etc/asterisk/chan_dahdi.conf that will include the
global settings
;
; Span 1: WCFXO/0 Wildcard X100P Board 1 (MASTER)
;;; line=1 WCFXO/0/0 FXSKS (In use)
signalling=fxs_ks
callerid=asreceived
group=0
context=from-pstn
channel = 1
callerid=
group=
context=default
I don't really know if these configs are all that matters to this problem.
Kernel clock/timer stuff:
zgrep RTC /proc/config.gz |grep -vP '^#'
CONFIG_RTC_LIB=m
CONFIG_RTC_CLASS=m
CONFIG_RTC_INTF_SYSFS=y
CONFIG_RTC_INTF_PROC=y
CONFIG_RTC_INTF_DEV=y
zgrep TIMER /proc/config.gz |grep -vP '^#'
CONFIG_TIMERFD=y
CONFIG_HPET_TIMER=y
CONFIG_X86_PM_TIMER=y
CONFIG_HANGCHECK_TIMER=y
CONFIG_SND_TIMER=m
# lsmod |grep rtc
genrtc 6092 0
# lsmod |grep timer
snd_timer 15428 2 snd_seq,snd_pcm
snd37348 11
snd_seq,snd_pcm_oss,snd_mixer_oss,snd_ens1371,snd_rawmidi,snd_seq_device,snd_ac97_codec,snd_pcm,snd_timer
Cheers,
--
|\ /|| | ~ ~
| \/ ||---| `|` ?
||ichael | |iggins\^ /
michael.higgins[at]evolone[dot]org
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Re: [asterisk-users] USA BRI -- any hope at all?
On Tue, 27 Jan 2009 20:35:56 -0500 Steve Totaro stot...@first-notification.com wrote: Laughing at Michael Higgins, ;-) Who said he wants HIS software to be Open Source? It is his and if he want to, God forbid, make a profit from his work, then who the hell are you to say it is not right? As I typed in the message to the list, Okay, fine. I don't insist that anyone do anything they don't want to do, believe me! This is why, in my OP to this list, I asked about tradition, are developers normally keeping modified code from the originators of the code? So, do I say it's not *right*? (I actually do understand the issues.) No. I say it's kinda crappy behaviour... ugh, from what I've seen so far. Maybe the source *was* released to the community and nobody cared. Well, that's still pretty lousy... perhaps deserving, I don't know. Yes, I have and will continue to speak with Marcin, and most of the prominent people involved with Asterisk in the past and currently. Is this a bad thing? I don't understand this... Did I indicate that you shouldn't? Who am I to say, anyway? What I did was try to see if Marcin had any posts to the mailing list. He doesn't, apparently, but rather I found in a post you mention that you had some conversation. Again, funny, that. What you did not see in that thread since it was taken off the list, man to man. Verizon BRI works with a off the shelf 1.4.x build and a BRI card somewhat. There are a great deal of missing IEs or Unknown IEs, caller ID does not work at all, when first starting the box, oubtbound calls work right away, then it takes about 30-40 minutes for inbound to start working. At random intervals, the card would stop taking calls for no reason and then start again after 30-40 minutes. If you can deal with that, then go for it. Of course I can't deal with that. It's broken! Why would I choose to deal with that?!? '-) The above were my final results with Verizon BRI service and 1.4.x. Hmm. If I wanted to stay with 1.4, apparently I could just buy the Quad(4) port BRI card from phoneicq and have to get all support directly from one person. It seems too risky, period. BRI is stupid in the US anyways, that is why it was never supported or proper stack written (with the exception of Marcin's). BRI was a miscarriage in the US and it will probably disappear in the near future. Okay, great! Why didn't you just reply with that to begin? Seriously...? '-) Look, this confirms what I'd pretty much concluded already. All I wanted to know. Now I can stop trying to find out what the deal is, you see. . . . Truly, I thought getting a public recommendation to contact someone who already hadn't passed the please don't taint my GNU/Linux and open source machine test, was, well, funny... okay? I found it *particularly* funny as you didn't address the part of my post where I want to know if it's common to, well... you can call it what you want with borrowing, and keeping modified, open source code (if that's what happened)... for me, I'm *not* a capitalist, but I do understand the effect on open source when folks do that. ;-) Just get a partial PRI, POTS, or a some DSL lines. Partial PRI is also disappearing, I've been told -- though now I'll have to try harder to find it since I don't need any bandwidth, but would like digital lines. POTS will likely have to do after all. I really appreciate your input. Obviously, you are very knowledgeable and helpful to solve problems. But I'm just looking for software to drive a card, didn't know what card to get, this is the software user list... As I said in my last post, I don't personally care about 'pissing matches'. It's not my money, I'm in no great hurry Why not be sure to add your statement about US BRI to a FAQ, or something? It seems germane... after all it takes some time and research just to realize 'BRI' isn't the same as 'BRI' in the US. Here, you've just definitively answered the question with, to paraphrase, that it's a dead end, at best... THANK YOU! '-) Cheers, -- |\ /|| | ~ ~ | \/ ||---| `|` ? ||ichael | |iggins\^ / michael.higgins[at]evolone[dot]org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] USA BRI -- any hope at all?
Folks -- First, apologies for not lurking for weeks or months to get the culture of the list. I read the recent post about improvement to the quality of posts with some amusement and full agreement. The problem is a big and very real one. I hope I'm not deepening it. But my question isn't explicitly asked with this subject line or definitively answered in the archives -- that I have found. What I did find left me with the impression that USA 'BRI', uh, '2B1Q' protocol(?) is not supported by *any* hardware vendor, at all, period, nor is it tested and proved in the software... stack(?), in one related branch or another on the OS side. A couple of direct inquiries to card vendors have dead-ended with a flat no, or requests for development funds(!) -- apparently there is code for one card, one vendor, that runs against 'bristuff', or did at one time, but wasn't maintained through several Asterisk releases (if the code was even released to the community... IDK). Is this common, that someone codes to their chip on their card and sells it to one or two consumers, then lets it drop and never gives the code up for continued development? (It seems contrary to GNU/Linux licensing conventions, but, again, I'm not paid as a software developer. I just think they might have sold more cards with a less proprietary approach.) Anyway, can I, with confidence, state (to the $employer) that Asterisk on linux via USA 'BRI' digital lines simply isn't possible? (In that, obviously, I can't pay for development nor do beta testing, each with vague hope that it might work okay someday...) If this is the case, then I must use multiple analog lines to access PSTN, or pay premium for 'PRI' pipes (80% of which we will never need)... is that about correct? Thanks in advance for any pointers, specific RTFM suggestions, any help appreciated. If there is a different list to post this query to, I'm not (yet) aware of it. Cheers, -- |\ /|| | ~ ~ | \/ ||---| `|` ? ||ichael | |iggins\^ / michael.higgins[at]evolone[dot]org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] USA BRI -- any hope at all?
On Tue, 27 Jan 2009 20:43:30 +0200 Tzafrir Cohen tzafrir.co...@xorcom.com wrote: To the best of my understanding, latest Asterisk should support it through chan_dahdi . Cool. Got any links to any related informations, so I can go by more than anonymous hearsay? '-) No need for extra bristuff or whatever. But this needs some testing. Ah. But I can't replace the business telephone system with a testing system for the company that pays me to implement working stuff, can I? Cheers, -- |\ /|| | ~ ~ | \/ ||---| `|` ? ||ichael | |iggins\^ / michael.higgins[at]evolone[dot]org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] USA BRI -- any hope at all?
On Tue, 27 Jan 2009 11:44:12 -0700 Wilton Helm wh...@compuserve.com wrote: I'm in the same boat and have been looking at this for several months, but haven't actually jumped in, hands-on, yet. No, I don't think the situation is as dismal as you paint it, although the lack of appropriate marketing for BRI in the US has all but killed it here, making it relatively unattractive to vendors. Right. This is what I've come to understand from the archives. I have been advised of several cards that support it, but they were two or four port cards that were serious overkill for my application, and I haven't found ONE card that supports it, so maybe we could compare notes? Rather, can you tell me who claims to support what hardware, so I can confirm? '-) expectedly pricy as a result. This isn't my $$ so I don't care about initial cost. I do care that it is proven, though. I am trying to use an HFC card, and have been advised that it is possible, but will take some digging and experimenting. I have no time for digging and experimenting, unfortunately. I expect that time will be taken up entirely trying to install and configure asterisk. As near as I have been able to figure out, mISDN is the appropriate place to begin, but I haven't tried to sort it out, yet. There are a couple of other paths that might lead to a solution, but they seem to be less supported than mISDN at present (if that is even possible). It sounds like Dahdi is moving towards that, but I don't think its quite there, yet. Okay, these are various driver protocols in the kernel/user space, right? Since I haven't purchased anything, should I be analysing the rate and support for development of these different protocols and informing my decision that way? Qwest is one of the few providers I know of that prices BRI reasonably. I have one up and running on a TA and use it for my voice service. As soon as I can figure out mISDN, I plan to move it to Asterisk. (I'll keep a TA around for backup). Hmm. Yes, Qwest 'BRI' was the intent... the problem is, any solution that doesn't lease equipment from the provider conflicts to their bottom-line interest, so I don't expect any help from them. Am I wrong in that? At any rate, if you could send along some info about cards that are purported to work here, I'd *really* appreciate it. '-) Thanks, -- |\ /|| | ~ ~ | \/ ||---| `|` ? ||ichael | |iggins\^ / michael.higgins[at]evolone[dot]org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] USA BRI -- any hope at all?
On Tue, 27 Jan 2009 12:56:46 -0500 Jon Pounder j...@inline.net wrote: I barked up the same tree you are barking for a while and just gave up - lots of you could buy this and try it, but no proven solution. That's exactly what I've come up with. Thanks for your reply. I don't see anything truly contradicting this in the other posts, so for now, it's doubtlessly a dead-end avenue... I do think it really is that bad, but I'd still like to be dissuaded of that. Cheers, -- |\ /|| | ~ ~ | \/ ||---| `|` ? ||ichael | |iggins\^ / michael.higgins[at]evolone[dot]org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] USA BRI -- any hope at all?
On Tue, 27 Jan 2009 12:27:04 -0600 Jerry Jones jjo...@danrj.com wrote: Instead you could always get a SIP/IAX provider. Can you please elaborate as to how this answers my question? Would getting a SIP/IAX provider be the same as getting a working 'BRI' ISDN line coming into the machine and properly handled in the chip or software stack? I know I can avoid all of this by paying a provider for services (which is what we do now). Is this proposed solution somehow different (other than cheaper) than that? I have COLO with fat bandwidth included, no PSTN lines. So, is this a viable alternative? How do I reliably fax from my server over SIP/IAX? Why would I pay for services that I have oodles of bandwidth to handle for a nominal investment? Example, you spend hours trying to get a tech support individual on the phone with any vendor, time where you learn nothing and get nowhere. Often, they don't know the answer anyway. Sometimes I call them back to tell them their business. OTOH, I can generally track down the patch, or library, or clue I need to fix anything open source and free, within an hour or two. Plus, I'm learning about the issues related and technologies underlying in the process. Anyway, I just don't see how this reply is at all helpful... say, if I intend to beta test asterisk and skype (I'm on the list, if they ever move forward), will this be possible on this kind of system? Would it be as useful, as flexible? Perhaps you were just being sarcastic -- since the lack of BRI sales (and marketing) *seems* to allow the opening to sales of these halfway services (where they wouldn't likely have much chance to sell them otherwise) -- sarcasm would make some sense. Unfortunately, in email sarcasm/sarcasm is the only way I've seen to reliably indicate communication as such. The terseness and offhandedness of the comment does inform toward this, but I don't really know... you. Nonetheless, I've 'bitten'. Let's see what you have to say. '-) Thanks for your reply. Cheers, -- |\ /|| | ~ ~ | \/ ||---| `|` ? ||ichael | |iggins\^ / michael.higgins[at]evolone[dot]org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] USA BRI -- any hope at all?
On Tue, 27 Jan 2009 16:55:02 -0500 Steve Totaro stot...@totarotechnologies.com wrote: On Tue, Jan 27, 2009 at 4:07 PM, Tzafrir Cohen tzafrir.co...@xorcom.com wrote: On Tue, Jan 27, 2009 at 03:45:34PM -0500, Steve Totaro wrote: Get a hold of Marcin Pyco (Former Digium Employee and extrememly smart guy. He has code/patches for zaptel to US BRIs work that include SPID as a variable in zap confs. Could you please expand on that point? Why should such patches remain secret? And are those patches really necessary now that chan_dahdi knows that there is such a thing called BRI? [8] BRISTUFF never worked properly with US BRI. [8] From what I understand it Marcin's code works on all card's that BRIStuff would work on, but does not because US BRI is a bit different than other BRI around the world. About keeping secrets, which he doesn't, he has said that he had these patches for a long time now. At any rate, it is his code to whatever he wants. If he wants to charge for it, do consulting, or keep it a secret, it is not your business nor do you have any say what he wants to do with his code. I think you just want your hands on the code since you work for a competitor but I may be wrong Oh, I'm laughing... Yes, Martin Pyco has already been in contact with me. What happened to open source community? Everyone who has these cards, IF THE PATCHES WORK, would benefit, no? But I'm not going to, not able to, invest, or risk, on one developer. Yes, he has said I can pay him... but that was his private email to me, not to the list. I haven't seen the old patches on the 'net for all to peruse (as I haven't really looked for them either). My reply, which *is* my business, however, I'll share here redacted, as it were: quote [...] I don't get this. You patched/borrowed/used bristuff code? Where is it? Is it any good, so it can it be ported to the current asterisk? [...] I don't understand. Either it works, or it doesn't. If you modified the code, where is it? Just because *you* abandoned it, doesn't mean it couldn't be maintained by someone else. Unless you don't get the GNU licensing thing... [...] Obviously, I'm trying to buy hardware which works with US '2B1Q' BRI. I don't buy software, except in those few cases where the chip is fixed already, like a win modem on a motherboard. If I got a free laptop with your chip in it, I might be persuaded to throw a few (100) bucks at the developers to be able to use it, but !...@#$%^*, man, you sell the card itself. You will sell more of them with better drivers (if the hardware is good to start, of course). Basically, if you want to sell the card, you have to show me, and everyone, the code. I can't possibly commit to supporting your coding unless it is released GPL, so others can spot errors, find security flaws, etc. Maybe I'm misunderstanding your reply, but it seems like you wrote something you didn't release to the community. No wonder you didn't sell many cards... way too risky to rely on just you. Sorry. /quoted Unbelievably (to me, anyway), he did reply, and, in fact, I did not misunderstand one bit. Standing his ground, stating, in rather large numbers, the market value of what I want to achieve, should I try without his code. IOW, implication is it will cost me *way* more to do it a different, available way, so... Okay, fine. I don't insist that anyone do anything they don't want to do, believe me! This is why, in my OP to this list, I asked about tradition, are developers normally keeping modified code from the originators of the code? But, I don't know what the actual license for the code in question is called, or what it allows, and finally, 'IANAL'. Just seems like, well, kinda crappy behaviour. Is this normal? I mean, sure, don't do it for me for free, but damn, man, let's see what you did to make it work... let someone else use it, test it, improve it, next thing you know there's a market for your hardware, where currently, there isn't. I mean, I'd take up a collection, or something, but still I'd be beta testing. '-) . . . Geez, I see that I'm replying to someone who appears to have some prior conversation with Mr Pyco as well. http://archives.free.net.ph/message/20081013.223907.3370db07.da.html Skribent: Steve Totaro Dato: 2008-10-13 15:39 -700 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: Re: [asterisk-users] ISDN ... unless this is some fluke, or something. Some guy with your name sez he can do this out of the box, more or less: Whatever the rub, using BRIStuff, Zaptel 1.4.X and a Junghanns' card or knock-off (and even Sangoma drivers), it will work with Verizon. Verizon is totally an option for me. So... where *is* that how-to posted? '-) I am still baffled. I don't personally care about pissing matches. It's not my money, I'm in no great hurry, and as someone pointed out, I could just pay the telco to do it for me. I will
