Re: [asterisk-users] Problem on Dialling-out
Bruce, Thanks. I already figured out the problem. It seems that a firewall issue. Regards, Malvin On 7/13/2011 12:30 PM, Bruce B wrote: Your trunk shows busy: */ -- Called CordiaVoIP/639285010430 -- SIP/CordiaVoIP-0015 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0)/* Try this in the CLI (asterisk -r): *core set verbose 0* *sip set debug peer CordiaVoIP* And then make a call and read why the SIP trunk is failing. -Bruce On Wed, Jul 13, 2011 at 12:23 AM, Malvin Rito mr...@mail.altcladding.com.ph mailto:mr...@mail.altcladding.com.ph wrote: Hi List, I have a Asterisk + FreePbx Server setup with around 10 SIP extensions and 1 VoIP trunk (CordiaVoIP), when we dial-out to any number call is being dropped with the following message on asterisk log: == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Called CordiaVoIP/639285010430 -- SIP/CordiaVoIP-0015 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Executing [s@macro-dialout-trunk:20] NoOp(SIP/1001-0014, Dial failed for some reason with DIALSTATUS = CONGESTION and HANGUPCAUSE = 0) in new stack -- Executing [s@macro-dialout-trunk:21] Goto(SIP/1001-0014, s-CONGESTION,1) in new stack -- Goto (macro-dialout-trunk,s-CONGESTION,1) -- Executing [s-CONGESTION@macro-dialout-trunk:1] Set(SIP/1001-0014, RC=0) in new stack -- Executing [s-CONGESTION@macro-dialout-trunk:2] Goto(SIP/1001-0014, 0,1) in new stack -- Goto (macro-dialout-trunk,0,1) -- Executing [0@macro-dialout-trunk:1] Goto(SIP/1001-0014, continue,1) in new stack -- Goto (macro-dialout-trunk,continue,1) -- Executing [continue@macro-dialout-trunk:1] GotoIf(SIP/1001-0014, 1?noreport) in new stack -- Goto (macro-dialout-trunk,continue,3) -- Executing [continue@macro-dialout-trunk:3] NoOp(SIP/1001-0014, TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 0 - failing through to other trunks) in new stack -- Executing [continue@macro-dialout-trunk:4] Set(SIP/1001-0014, CALLERID(number)=1001) in new stack -- Executing [639285010430@from-internal:8] Macro(SIP/1001-0014, outisbusy,) in new stack -- Executing [s@macro-outisbusy:1] Progress(SIP/1001-0014, ) in new stack -- Executing [s@macro-outisbusy:2] Playback(SIP/1001-0014, all-circuits-busy-now,noanswer) in new stack -- SIP/1001-0014 Playing 'all-circuits-busy-now.gsm' (language 'en') -- Executing [s@macro-outisbusy:3] Playback(SIP/1001-0014, pls-try-call-later,noanswer) in new stack -- SIP/1001-0014 Playing 'pls-try-call-later.gsm' (language 'en') -- Executing [s@macro-outisbusy:4] Macro(SIP/1001-0014, hangupcall) in new stack -- Executing [s@macro-hangupcall:1] GotoIf(SIP/1001-0014, 1?skiprg) in new stack -- Goto (macro-hangupcall,s,4) -- Executing [s@macro-hangupcall:4] GotoIf(SIP/1001-0014, 1?skipblkvm) in new stack -- Goto (macro-hangupcall,s,7) -- Executing [s@macro-hangupcall:7] GotoIf(SIP/1001-0014, 1?theend) in new stack -- Goto (macro-hangupcall,s,9) -- Executing [s@macro-hangupcall:9] Hangup(SIP/1001-0014, ) in new stack == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/1001-0014' in macro 'hangupcall' == Spawn extension (macro-outisbusy, s, 4) exited non-zero on 'SIP/1001-0014' in macro 'outisbusy' == Spawn extension (from-internal, 639285010430, 8) exited non-zero on 'SIP/1001-0014' -- Executing [h@from-internal:1] Macro(SIP/1001-0014, hangupcall) in new stack -- Executing [s@macro-hangupcall:1] GotoIf(SIP/1001-0014, 1?skiprg) in new stack -- Goto (macro-hangupcall,s,4) -- Executing [s@macro-hangupcall:4] GotoIf(SIP/1001-0014, 1?skipblkvm) in new stack -- Goto (macro-hangupcall,s,7) -- Executing [s@macro-hangupcall:7] GotoIf(SIP/1001-0014, 1?theend) in new stack -- Goto (macro-hangupcall,s,9) -- Executing [s@macro-hangupcall:9] Hangup(SIP/1001-0014, ) in new stack == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/1001-0014' in macro 'hangupcall' == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/1001-0014' localhost*CLI Can someone assist me please. Thanks in advance. Regards, Malvin -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit:
[asterisk-users] Connect Avaya to Asterisk PBX
Hi List,
I have another issue on allowing outgoing calls to PSTN on Asterisk via
Avaya Phones, I hope that anyone could help me fix this issue:
*When I dial through Avaya phone i just here a good bye message reply
from asterisk server. And here is the log:*
== Starting OOH323/(null)-b7db8aa0 at internal,s,1 failed so falling
back to exten 's'
== Starting OOH323/(null)-b7db8aa0 at internal,s,1 still failed so
falling back to context 'default'
-- Executing [s@default:1] Playback(OOH323/(null)-b7db8aa0,
vm-goodbye) in new stack
-- OOH323/(null)-b7db8aa0 Playing 'vm-goodbye.ulaw' (language 'en')
-- Executing [s@default:2] Macro(OOH323/(null)-b7db8aa0,
hangupcall) in new stack
-- Executing [s@macro-hangupcall:1]
GotoIf(OOH323/(null)-b7db8aa0, 1?skiprg) in new stack
-- Goto (macro-hangupcall,s,4)
-- Executing [s@macro-hangupcall:4]
GotoIf(OOH323/(null)-b7db8aa0, 1?skipblkvm) in new stack
-- Goto (macro-hangupcall,s,7)
-- Executing [s@macro-hangupcall:7]
GotoIf(OOH323/(null)-b7db8aa0, 1?theend) in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [s@macro-hangupcall:9]
Hangup(OOH323/(null)-b7db8aa0, ) in new stack
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on
'OOH323/(null)-b7db8aa0' in macro 'hangupcall'
== Spawn extension (default, s, 2) exited non-zero on
'OOH323/(null)-b7db8aa0'
-- Executing [h@default:1] Macro(OOH323/(null)-b7db8aa0,
hangupcall,) in new stack
-- Executing [s@macro-hangupcall:1]
GotoIf(OOH323/(null)-b7db8aa0, 1?skiprg) in new stack
-- Goto (macro-hangupcall,s,4)
-- Executing [s@macro-hangupcall:4]
GotoIf(OOH323/(null)-b7db8aa0, 1?skipblkvm) in new stack
-- Goto (macro-hangupcall,s,7)
-- Executing [s@macro-hangupcall:7]
GotoIf(OOH323/(null)-b7db8aa0, 1?theend) in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [s@macro-hangupcall:9]
Hangup(OOH323/(null)-b7db8aa0, ) in new stack
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on
'OOH323/(null)-b7db8aa0' in macro 'hangupcall'
== Spawn extension (default, h, 1) exited non-zero on
'OOH323/(null)-b7db8aa0'
*Here is also the content of my extensions_custom.conf:*
[general]
static=yes
autofallthrough=yes
[internal]
exten = 1000,1,Dial(SIP/1000)
exten = 1000,2,HangUp()
exten = _,1,Dial(H323/${EXTEN}@Avaya)
exten = _XXX,1,Dial(H323/${EXTEN}@Avaya)
exten = _XX,1,Dial(H323/${EXTEN}@Avaya)
*Here is also the content of my ooh323.conf:*
[general]
faststart=yes
h245tunneling=yes
gatekeeper=DISABLE
bindaddr=10.1.129.231
port=1720
callerID=ALT Asterisk PBX
progress_setup=8
progress_alert=8
disallow=all
allow=all
dtmfmode=inband
faststart=yes
context=internal
[Avaya]
type=friend
context=internal
host=10.1.129.247
port=1720
canreinvite=no
disallow=all
allow=alaw
dtmfmode=inband
*Here is also the content of sip_custom.conf:*
[general]
context=internal
videosupport=yes
allow=h261
allow=h263
allow=h263p
bindaddr=10.1.129.231
srvlookup=yes
conreinvitte=no
[1000]
type=friend
secret=malvin123
host=dynamic
dtmfmode=inband
disallow=all
allow=all
nat=yes
Thanks regards,
Malvin
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Re: [asterisk-users] Connect Avaya to Asterisk PBX
Looks like you need an 's' exten in your [internal] context.
Thanks,
--Warren Selby, dCAP
On Jul 13, 2011, at 2:02 AM, Malvin Rito mr...@mail.altcladding.com.ph wrote:
Hi List,
I have another issue on allowing outgoing calls to PSTN on Asterisk via Avaya
Phones, I hope that anyone could help me fix this issue:
When I dial through Avaya phone i just here a good bye message reply from
asterisk server. And here is the log:
== Starting OOH323/(null)-b7db8aa0 at internal,s,1 failed so falling back to
exten 's'
== Starting OOH323/(null)-b7db8aa0 at internal,s,1 still failed so
falling back to context 'default'
-- Executing [s@default:1] Playback(OOH323/(null)-b7db8aa0,
vm-goodbye) in new stack
-- OOH323/(null)-b7db8aa0 Playing 'vm-goodbye.ulaw' (language 'en')
-- Executing [s@default:2] Macro(OOH323/(null)-b7db8aa0, hangupcall)
in new stack
-- Executing [s@macro-hangupcall:1] GotoIf(OOH323/(null)-b7db8aa0,
1?skiprg) in new stack
-- Goto (macro-hangupcall,s,4)
-- Executing [s@macro-hangupcall:4] GotoIf(OOH323/(null)-b7db8aa0,
1?skipblkvm) in new stack
-- Goto (macro-hangupcall,s,7)
-- Executing [s@macro-hangupcall:7] GotoIf(OOH323/(null)-b7db8aa0,
1?theend) in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [s@macro-hangupcall:9] Hangup(OOH323/(null)-b7db8aa0, )
in new stack
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on
'OOH323/(null)-b7db8aa0' in macro 'hangupcall'
== Spawn extension (default, s, 2) exited non-zero on
'OOH323/(null)-b7db8aa0'
-- Executing [h@default:1] Macro(OOH323/(null)-b7db8aa0, hangupcall,)
in new stack
-- Executing [s@macro-hangupcall:1] GotoIf(OOH323/(null)-b7db8aa0,
1?skiprg) in new stack
-- Goto (macro-hangupcall,s,4)
-- Executing [s@macro-hangupcall:4] GotoIf(OOH323/(null)-b7db8aa0,
1?skipblkvm) in new stack
-- Goto (macro-hangupcall,s,7)
-- Executing [s@macro-hangupcall:7] GotoIf(OOH323/(null)-b7db8aa0,
1?theend) in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [s@macro-hangupcall:9] Hangup(OOH323/(null)-b7db8aa0, )
in new stack
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on
'OOH323/(null)-b7db8aa0' in macro 'hangupcall'
== Spawn extension (default, h, 1) exited non-zero on
'OOH323/(null)-b7db8aa0'
Here is also the content of my extensions_custom.conf:
[general]
static=yes
autofallthrough=yes
[internal]
exten = 1000,1,Dial(SIP/1000)
exten = 1000,2,HangUp()
exten = _,1,Dial(H323/${EXTEN}@Avaya)
exten = _XXX,1,Dial(H323/${EXTEN}@Avaya)
exten = _XX,1,Dial(H323/${EXTEN}@Avaya)
Here is also the content of my ooh323.conf:
[general]
faststart=yes
h245tunneling=yes
gatekeeper=DISABLE
bindaddr=10.1.129.231
port=1720
callerID=ALT Asterisk PBX
progress_setup=8
progress_alert=8
disallow=all
allow=all
dtmfmode=inband
faststart=yes
context=internal
[Avaya]
type=friend
context=internal
host=10.1.129.247
port=1720
canreinvite=no
disallow=all
allow=alaw
dtmfmode=inband
Here is also the content of sip_custom.conf:
[general]
context=internal
videosupport=yes
allow=h261
allow=h263
allow=h263p
bindaddr=10.1.129.231
srvlookup=yes
conreinvitte=no
[1000]
type=friend
secret=malvin123
host=dynamic
dtmfmode=inband
disallow=all
allow=all
nat=yes
Thanks regards,
Malvin
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Re: [asterisk-users] Connect Avaya to Asterisk PBX
How do I write it on my code?
On 7/13/2011 4:04 PM, Warren Selby wrote:
Looks like you need an 's' exten in your [internal] context.
Thanks,
--Warren Selby, dCAP
On Jul 13, 2011, at 2:02 AM, Malvin Rito
mr...@mail.altcladding.com.ph mailto:mr...@mail.altcladding.com.ph
wrote:
Hi List,
I have another issue on allowing outgoing calls to PSTN on Asterisk
via Avaya Phones, I hope that anyone could help me fix this issue:
*When I dial through Avaya phone i just here a good bye message
reply from asterisk server. And here is the log:*
== Starting OOH323/(null)-b7db8aa0 at internal,s,1 failed so falling
back to exten 's'
== Starting OOH323/(null)-b7db8aa0 at internal,s,1 still failed so
falling back to context 'default'
-- Executing [s@default:1] Playback(OOH323/(null)-b7db8aa0,
vm-goodbye) in new stack
-- OOH323/(null)-b7db8aa0 Playing 'vm-goodbye.ulaw' (language 'en')
-- Executing [s@default:2] Macro(OOH323/(null)-b7db8aa0,
hangupcall) in new stack
-- Executing [s@macro-hangupcall:1]
GotoIf(OOH323/(null)-b7db8aa0, 1?skiprg) in new stack
-- Goto (macro-hangupcall,s,4)
-- Executing [s@macro-hangupcall:4]
GotoIf(OOH323/(null)-b7db8aa0, 1?skipblkvm) in new stack
-- Goto (macro-hangupcall,s,7)
-- Executing [s@macro-hangupcall:7]
GotoIf(OOH323/(null)-b7db8aa0, 1?theend) in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [s@macro-hangupcall:9]
Hangup(OOH323/(null)-b7db8aa0, ) in new stack
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on
'OOH323/(null)-b7db8aa0' in macro 'hangupcall'
== Spawn extension (default, s, 2) exited non-zero on
'OOH323/(null)-b7db8aa0'
-- Executing [h@default:1] Macro(OOH323/(null)-b7db8aa0,
hangupcall,) in new stack
-- Executing [s@macro-hangupcall:1]
GotoIf(OOH323/(null)-b7db8aa0, 1?skiprg) in new stack
-- Goto (macro-hangupcall,s,4)
-- Executing [s@macro-hangupcall:4]
GotoIf(OOH323/(null)-b7db8aa0, 1?skipblkvm) in new stack
-- Goto (macro-hangupcall,s,7)
-- Executing [s@macro-hangupcall:7]
GotoIf(OOH323/(null)-b7db8aa0, 1?theend) in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [s@macro-hangupcall:9]
Hangup(OOH323/(null)-b7db8aa0, ) in new stack
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on
'OOH323/(null)-b7db8aa0' in macro 'hangupcall'
== Spawn extension (default, h, 1) exited non-zero on
'OOH323/(null)-b7db8aa0'
*Here is also the content of my extensions_custom.conf:*
[general]
static=yes
autofallthrough=yes
[internal]
exten = 1000,1,Dial(SIP/1000)
exten = 1000,2,HangUp()
exten = _,1,Dial(H323/${EXTEN}@Avaya)
exten = _XXX,1,Dial(H323/${EXTEN}@Avaya)
exten = _XX,1,Dial(H323/${EXTEN}@Avaya)
*Here is also the content of my ooh323.conf:*
[general]
faststart=yes
h245tunneling=yes
gatekeeper=DISABLE
bindaddr=10.1.129.231
port=1720
callerID=ALT Asterisk PBX
progress_setup=8
progress_alert=8
disallow=all
allow=all
dtmfmode=inband
faststart=yes
context=internal
[Avaya]
type=friend
context=internal
host=10.1.129.247
port=1720
canreinvite=no
disallow=all
allow=alaw
dtmfmode=inband
*Here is also the content of sip_custom.conf:*
[general]
context=internal
videosupport=yes
allow=h261
allow=h263
allow=h263p
bindaddr=10.1.129.231
srvlookup=yes
conreinvitte=no
[1000]
type=friend
secret=malvin123
host=dynamic
dtmfmode=inband
disallow=all
allow=all
nat=yes
Thanks regards,
Malvin
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Re: [asterisk-users] Connect Avaya to Asterisk PBX
you can edit dial-plan by adding following lines to your code
[internal]
exten = s,1,Dial(SIP/1000)
exten = s,2,HangUp()
exten = 1000,1,Dial(SIP/1000)
exten = 1000,2,HangUp()
exten = _,1,Dial(H323/${EXTEN}@
Avaya)
exten = _XXX,1,Dial(H323/${EXTEN}@Avaya)
exten = _XX,1,Dial(H323/${EXTEN}@Avaya)
On Wed, Jul 13, 2011 at 1:35 PM, Malvin Rito
mr...@mail.altcladding.com.phwrote:
**
How do I write it on my code?
On 7/13/2011 4:04 PM, Warren Selby wrote:
Looks like you need an 's' exten in your [internal] context.
Thanks,
--Warren Selby, dCAP
On Jul 13, 2011, at 2:02 AM, Malvin Rito mr...@mail.altcladding.com.ph
wrote:
Hi List,
I have another issue on allowing outgoing calls to PSTN on Asterisk via
Avaya Phones, I hope that anyone could help me fix this issue:
*When I dial through Avaya phone i just here a good bye message reply
from asterisk server. And here is the log:*
== Starting OOH323/(null)-b7db8aa0 at internal,s,1 failed so falling back
to exten 's'
== Starting OOH323/(null)-b7db8aa0 at internal,s,1 still failed so
falling back to context 'default'
-- Executing [s@default:1] Playback(OOH323/(null)-b7db8aa0,
vm-goodbye) in new stack
-- OOH323/(null)-b7db8aa0 Playing 'vm-goodbye.ulaw' (language 'en')
-- Executing [s@default:2] Macro(OOH323/(null)-b7db8aa0,
hangupcall) in new stack
-- Executing [s@macro-hangupcall:1] GotoIf(OOH323/(null)-b7db8aa0,
1?skiprg) in new stack
-- Goto (macro-hangupcall,s,4)
-- Executing [s@macro-hangupcall:4] GotoIf(OOH323/(null)-b7db8aa0,
1?skipblkvm) in new stack
-- Goto (macro-hangupcall,s,7)
-- Executing [s@macro-hangupcall:7] GotoIf(OOH323/(null)-b7db8aa0,
1?theend) in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [s@macro-hangupcall:9] Hangup(OOH323/(null)-b7db8aa0,
) in new stack
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on
'OOH323/(null)-b7db8aa0' in macro 'hangupcall'
== Spawn extension (default, s, 2) exited non-zero on
'OOH323/(null)-b7db8aa0'
-- Executing [h@default:1] Macro(OOH323/(null)-b7db8aa0,
hangupcall,) in new stack
-- Executing [s@macro-hangupcall:1] GotoIf(OOH323/(null)-b7db8aa0,
1?skiprg) in new stack
-- Goto (macro-hangupcall,s,4)
-- Executing [s@macro-hangupcall:4] GotoIf(OOH323/(null)-b7db8aa0,
1?skipblkvm) in new stack
-- Goto (macro-hangupcall,s,7)
-- Executing [s@macro-hangupcall:7] GotoIf(OOH323/(null)-b7db8aa0,
1?theend) in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [s@macro-hangupcall:9] Hangup(OOH323/(null)-b7db8aa0,
) in new stack
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on
'OOH323/(null)-b7db8aa0' in macro 'hangupcall'
== Spawn extension (default, h, 1) exited non-zero on
'OOH323/(null)-b7db8aa0'
*Here is also the content of my extensions_custom.conf:*
[general]
static=yes
autofallthrough=yes
[internal]
exten = 1000,1,Dial(SIP/1000)
exten = 1000,2,HangUp()
exten = _,1,Dial(H323/${EXTEN}@Avaya)
exten = _XXX,1,Dial(H323/${EXTEN}@Avaya)
exten = _XX,1,Dial(H323/${EXTEN}@Avaya)
*Here is also the content of my ooh323.conf:*
[general]
faststart=yes
h245tunneling=yes
gatekeeper=DISABLE
bindaddr=10.1.129.231
port=1720
callerID=ALT Asterisk PBX
progress_setup=8
progress_alert=8
disallow=all
allow=all
dtmfmode=inband
faststart=yes
context=internal
[Avaya]
type=friend
context=internal
host=10.1.129.247
port=1720
canreinvite=no
disallow=all
allow=alaw
dtmfmode=inband
*Here is also the content of sip_custom.conf:*
[general]
context=internal
videosupport=yes
allow=h261
allow=h263
allow=h263p
bindaddr=10.1.129.231
srvlookup=yes
conreinvitte=no
[1000]
type=friend
secret=malvin123
host=dynamic
dtmfmode=inband
disallow=all
allow=all
nat=yes
Thanks regards,
Malvin
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Re: [asterisk-users] SoftHangup on asterisk 1.8.3.2 (renamed)
On Tue, 2011-07-12 at 09:13 +0100, Ishfaq Malik wrote:
On Thu, 2011-07-07 at 14:23 -0400, Jeremy Kister wrote:
On 7/7/2011 9:32 AM, Ishfaq Malik wrote:
I'm having the same issue on 1.8.3.2 (with a couple of patches)
exten = s,1,Set(CHAN=${SHELL(asterisk -rx core show channels | awk
'/^SIP\/vgw1-/ { print $1 }' | head -1)})
This turned out to be a PEBKAC error. A newline was attached to the
$CHAN variable.
adding | tr -d '\n' to the end of the command fixed it right up.
Well in that case I'm having a different issue.
When I do
channel request hangup SIP/-1136
I get a
Requested Hangup on channel 'SIP/-1136'
response but the channel never hangs up
I'm having to restart the asterisk to clear the channels and that is not
an optimum solution!
Has anyone else encountered this or can see something obvious that I'm
doing wrong?
Worked out what was happening. I was trying to hangup stale channels. As
a stale channel is not being written to or read from the hangup will
never execute.
--
Ishfaq Malik
Software Developer
PackNet Ltd
Office: 0161 660 3062
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[asterisk-users] How to Hang up a stale SIP channel?
Hi We're using asterisk 1.8.3.2 and are finding incidences of stale channels remaining after both parties have hung up. We have tried to hang the channel up using channel request hangup But by it's definition, it will not work as it only executes the hangup as soon as the the channel is written to or read from but as the channel is stale, it will not be written to or read from so the command will not instigate the hangup. Does anyone know of any way we can hangup a stale channel via the console? Thanks in Advance Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Connect Avaya to Asterisk PBX
Thanks. I want to dial-out to PSTN using Asterisk Server via Avaya Phone
using Cordia VoIP Service provider. How can I achieve it using the same
code below?
Regards,
Malvin
On 7/13/2011 4:59 PM, DHAVAL INDRODIYA wrote:
you can edit dial-plan by adding following lines to your code
[internal]
exten = s,1,Dial(SIP/1000)
exten = s,2,HangUp()
exten = 1000,1,Dial(SIP/1000)
exten = 1000,2,HangUp()
exten = _,1,Dial(H323/${EXTEN}@
Avaya)
exten = _XXX,1,Dial(H323/${EXTEN}@Avaya)
exten = _XX,1,Dial(H323/${EXTEN}@Avaya)
On Wed, Jul 13, 2011 at 1:35 PM, Malvin Rito
mr...@mail.altcladding.com.ph mailto:mr...@mail.altcladding.com.ph
wrote:
How do I write it on my code?
On 7/13/2011 4:04 PM, Warren Selby wrote:
Looks like you need an 's' exten in your [internal] context.
Thanks,
--Warren Selby, dCAP
On Jul 13, 2011, at 2:02 AM, Malvin Rito
mr...@mail.altcladding.com.ph
mailto:mr...@mail.altcladding.com.ph wrote:
Hi List,
I have another issue on allowing outgoing calls to PSTN on
Asterisk via Avaya Phones, I hope that anyone could help me fix
this issue:
*When I dial through Avaya phone i just here a good bye
message reply from asterisk server. And here is the log:*
== Starting OOH323/(null)-b7db8aa0 at internal,s,1 failed so
falling back to exten 's'
== Starting OOH323/(null)-b7db8aa0 at internal,s,1 still
failed so falling back to context 'default'
-- Executing [s@default:1]
Playback(OOH323/(null)-b7db8aa0, vm-goodbye) in new stack
-- OOH323/(null)-b7db8aa0 Playing 'vm-goodbye.ulaw'
(language 'en')
-- Executing [s@default:2] Macro(OOH323/(null)-b7db8aa0,
hangupcall) in new stack
-- Executing [s@macro-hangupcall:1]
GotoIf(OOH323/(null)-b7db8aa0, 1?skiprg) in new stack
-- Goto (macro-hangupcall,s,4)
-- Executing [s@macro-hangupcall:4]
GotoIf(OOH323/(null)-b7db8aa0, 1?skipblkvm) in new stack
-- Goto (macro-hangupcall,s,7)
-- Executing [s@macro-hangupcall:7]
GotoIf(OOH323/(null)-b7db8aa0, 1?theend) in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [s@macro-hangupcall:9]
Hangup(OOH323/(null)-b7db8aa0, ) in new stack
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on
'OOH323/(null)-b7db8aa0' in macro 'hangupcall'
== Spawn extension (default, s, 2) exited non-zero on
'OOH323/(null)-b7db8aa0'
-- Executing [h@default:1] Macro(OOH323/(null)-b7db8aa0,
hangupcall,) in new stack
-- Executing [s@macro-hangupcall:1]
GotoIf(OOH323/(null)-b7db8aa0, 1?skiprg) in new stack
-- Goto (macro-hangupcall,s,4)
-- Executing [s@macro-hangupcall:4]
GotoIf(OOH323/(null)-b7db8aa0, 1?skipblkvm) in new stack
-- Goto (macro-hangupcall,s,7)
-- Executing [s@macro-hangupcall:7]
GotoIf(OOH323/(null)-b7db8aa0, 1?theend) in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [s@macro-hangupcall:9]
Hangup(OOH323/(null)-b7db8aa0, ) in new stack
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on
'OOH323/(null)-b7db8aa0' in macro 'hangupcall'
== Spawn extension (default, h, 1) exited non-zero on
'OOH323/(null)-b7db8aa0'
*Here is also the content of my extensions_custom.conf:*
[general]
static=yes
autofallthrough=yes
[internal]
exten = 1000,1,Dial(SIP/1000)
exten = 1000,2,HangUp()
exten = _,1,Dial(H323/${EXTEN}@Avaya)
exten = _XXX,1,Dial(H323/${EXTEN}@Avaya)
exten = _XX,1,Dial(H323/${EXTEN}@Avaya)
*Here is also the content of my ooh323.conf:*
[general]
faststart=yes
h245tunneling=yes
gatekeeper=DISABLE
bindaddr=10.1.129.231
port=1720
callerID=ALT Asterisk PBX
progress_setup=8
progress_alert=8
disallow=all
allow=all
dtmfmode=inband
faststart=yes
context=internal
[Avaya]
type=friend
context=internal
host=10.1.129.247
port=1720
canreinvite=no
disallow=all
allow=alaw
dtmfmode=inband
*Here is also the content of sip_custom.conf:*
[general]
context=internal
videosupport=yes
allow=h261
allow=h263
allow=h263p
bindaddr=10.1.129.231
srvlookup=yes
conreinvitte=no
[1000]
type=friend
secret=malvin123
host=dynamic
dtmfmode=inband
disallow=all
allow=all
nat=yes
Thanks regards,
Malvin
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Re: [asterisk-users] Queue Issue : Duration between 2 agents call
Hy, I still struggle with this issue, does anybody can help me? Regards Le 10/07/2011 13:04, Florent THOMAS a écrit : Hy, I'm currently working with one queue and whatever I change in the config, it stills a gap of 6 seconds during which no agents are ringing for this queue. Is ther any parameter to configure there? regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to Hang up a stale SIP channel?
On Wed, Jul 13, 2011 at 5:35 AM, Ishfaq Malik i...@pack-net.co.uk wrote: Hi We're using asterisk 1.8.3.2 and are finding incidences of stale channels remaining after both parties have hung up. We have tried to hang the channel up using channel request hangup But by it's definition, it will not work as it only executes the hangup as soon as the the channel is written to or read from but as the channel is stale, it will not be written to or read from so the command will not instigate the hangup. Does anyone know of any way we can hangup a stale channel via the console? I've had this happen a few times, but with 1.6.2. I ended up writing .call files to the asterisk spool directory instructing it to hang up a particular sip channel. -M -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDRs
Trying to create and populate arbitrary column in cdr table in mysql database. I created column ‘hangupcause’ in cdr table and setting CDR(hangupcause) variable on h extension, but database column in not getting populated. It is showing below in logs though: Executing [h@from-pstn:2] Set(SIP/abc, CDR(hangupcause)=16) in new stack Any idea why is that? On 12 July 2011 16:33, deeps backup backup.de...@gmail.com wrote: Hi Like we can define cdr field format for csv, is it possible to define if cdrs are stored in a database? Also, what will be size limit for database CDR storage ? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Mysterious dropped calls
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mark Rosedale Sent: Tuesday, July 12, 2011 4:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Mysterious dropped calls So I'm now using asterisk 1.8.5rc1 for Asterisk. I'm still getting mysterious dropped calls. This only happens on calls that are outbound on Dahdi and mostly happens in conference calls particularly 8xx-xxx- This is the output of the hangup. [Ksebpbx1*CLI [0KPRI Span: 1 q931_hangup: other hangup PRI Span: 1 NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Active, busydetect=yes or callprogress=yes in chan_dahdi.conf often cause random call hangups. If you have those options set, either remove them or set them to no. Can you elaborate on that? I have callprogress=yes active. Is this a known bug or just a function of the option? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Benchmarking AGI performance in C, PHP, and Perl
Kevin P. Fleming kpflem...@digium.com writes: OT: Take a look at 'systemd'; this is exactly what's happening there, and Fedora is likely to incorporate it into Fedora 16, and it will make its way into other distros after that. It was incorporated into Fedora 14, and it is the default in Fedora 15... /Benny (And yes it meant I couldn't boot after upgrading to Fedora 15. It couldn't handle that I had the cgroup file system mounted on /cgroup in fstab.) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Mysterious dropped calls
Sent from a computer -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mark Rosedale Sent: Wednesday, July 13, 2011 10:59 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Mysterious dropped calls -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mark Rosedale Sent: Tuesday, July 12, 2011 4:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Mysterious dropped calls So I'm now using asterisk 1.8.5rc1 for Asterisk. I'm still getting mysterious dropped calls. This only happens on calls that are outbound on Dahdi and mostly happens in conference calls particularly 8xx-xxx- This is the output of the hangup. [Ksebpbx1*CLI [0KPRI Span: 1 q931_hangup: other hangup PRI Span: 1 NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Active, busydetect=yes or callprogress=yes in chan_dahdi.conf often cause random call hangups. If you have those options set, either remove them or set them to no. Can you elaborate on that? I have callprogress=yes active. Is this a known bug or just a function of the option? 1) callprogress= is only useful on analog lines 2) per chan_dahdi.conf.sample This feature can also easily detect false hangups. The symptoms of this is being disconnected in the middle of a call for no reason. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Mysterious dropped calls
I'll change this immediately thanks, mjr On Jul 13, 2011, at 11:08 AM, Eric Wieling wrote: Sent from a computer -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mark Rosedale Sent: Wednesday, July 13, 2011 10:59 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Mysterious dropped calls -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mark Rosedale Sent: Tuesday, July 12, 2011 4:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Mysterious dropped calls So I'm now using asterisk 1.8.5rc1 for Asterisk. I'm still getting mysterious dropped calls. This only happens on calls that are outbound on Dahdi and mostly happens in conference calls particularly 8xx-xxx- This is the output of the hangup. [Ksebpbx1*CLI [0KPRI Span: 1 q931_hangup: other hangup PRI Span: 1 NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Active, busydetect=yes or callprogress=yes in chan_dahdi.conf often cause random call hangups. If you have those options set, either remove them or set them to no. Can you elaborate on that? I have callprogress=yes active. Is this a known bug or just a function of the option? 1) callprogress= is only useful on analog lines 2) per chan_dahdi.conf.sample This feature can also easily detect false hangups. The symptoms of this is being disconnected in the middle of a call for no reason. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem on Dialling-out
Yes, that is it. And you were inviting the provider to contact you back at your private subnet of 172.16.x.x: *From: Cordia sip:Unknown@172.16.9.15;tag=**as2267fdcc* * * So, hence their responces never made it back to you and that's why you are re-transmitting 6 times to get attention. * * - Bruce On Wed, Jul 13, 2011 at 2:49 AM, Malvin Rito mr...@mail.altcladding.com.phwrote: ** Bruce, Thanks. I already figured out the problem. It seems that a firewall issue. Regards, Malvin On 7/13/2011 12:30 PM, Bruce B wrote: Your trunk shows busy: * -- Called CordiaVoIP/639285010430 -- SIP/CordiaVoIP-0015 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0)* Try this in the CLI (asterisk -r): *core set verbose 0* *sip set debug peer CordiaVoIP* And then make a call and read why the SIP trunk is failing. -Bruce On Wed, Jul 13, 2011 at 12:23 AM, Malvin Rito mr...@mail.altcladding.com.ph wrote: Hi List, I have a Asterisk + FreePbx Server setup with around 10 SIP extensions and 1 VoIP trunk (CordiaVoIP), when we dial-out to any number call is being dropped with the following message on asterisk log: == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Called CordiaVoIP/639285010430 -- SIP/CordiaVoIP-0015 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Executing [s@macro-dialout-trunk:20] NoOp(SIP/1001-0014, Dial failed for some reason with DIALSTATUS = CONGESTION and HANGUPCAUSE = 0) in new stack -- Executing [s@macro-dialout-trunk:21] Goto(SIP/1001-0014, s-CONGESTION,1) in new stack -- Goto (macro-dialout-trunk,s-CONGESTION,1) -- Executing [s-CONGESTION@macro-dialout-trunk:1] Set(SIP/1001-0014, RC=0) in new stack -- Executing [s-CONGESTION@macro-dialout-trunk:2] Goto(SIP/1001-0014, 0,1) in new stack -- Goto (macro-dialout-trunk,0,1) -- Executing [0@macro-dialout-trunk:1] Goto(SIP/1001-0014, continue,1) in new stack -- Goto (macro-dialout-trunk,continue,1) -- Executing [continue@macro-dialout-trunk:1] GotoIf(SIP/1001-0014, 1?noreport) in new stack -- Goto (macro-dialout-trunk,continue,3) -- Executing [continue@macro-dialout-trunk:3] NoOp(SIP/1001-0014, TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 0 - failing through to other trunks) in new stack -- Executing [continue@macro-dialout-trunk:4] Set(SIP/1001-0014, CALLERID(number)=1001) in new stack -- Executing [639285010430@from-internal:8] Macro(SIP/1001-0014, outisbusy,) in new stack -- Executing [s@macro-outisbusy:1] Progress(SIP/1001-0014, ) in new stack -- Executing [s@macro-outisbusy:2] Playback(SIP/1001-0014, all-circuits-busy-now,noanswer) in new stack -- SIP/1001-0014 Playing 'all-circuits-busy-now.gsm' (language 'en') -- Executing [s@macro-outisbusy:3] Playback(SIP/1001-0014, pls-try-call-later,noanswer) in new stack -- SIP/1001-0014 Playing 'pls-try-call-later.gsm' (language 'en') -- Executing [s@macro-outisbusy:4] Macro(SIP/1001-0014, hangupcall) in new stack -- Executing [s@macro-hangupcall:1] GotoIf(SIP/1001-0014, 1?skiprg) in new stack -- Goto (macro-hangupcall,s,4) -- Executing [s@macro-hangupcall:4] GotoIf(SIP/1001-0014, 1?skipblkvm) in new stack -- Goto (macro-hangupcall,s,7) -- Executing [s@macro-hangupcall:7] GotoIf(SIP/1001-0014, 1?theend) in new stack -- Goto (macro-hangupcall,s,9) -- Executing [s@macro-hangupcall:9] Hangup(SIP/1001-0014, ) in new stack == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/1001-0014' in macro 'hangupcall' == Spawn extension (macro-outisbusy, s, 4) exited non-zero on 'SIP/1001-0014' in macro 'outisbusy' == Spawn extension (from-internal, 639285010430, 8) exited non-zero on 'SIP/1001-0014' -- Executing [h@from-internal:1] Macro(SIP/1001-0014, hangupcall) in new stack -- Executing [s@macro-hangupcall:1] GotoIf(SIP/1001-0014, 1?skiprg) in new stack -- Goto (macro-hangupcall,s,4) -- Executing [s@macro-hangupcall:4] GotoIf(SIP/1001-0014, 1?skipblkvm) in new stack -- Goto (macro-hangupcall,s,7) -- Executing [s@macro-hangupcall:7] GotoIf(SIP/1001-0014, 1?theend) in new stack -- Goto (macro-hangupcall,s,9) -- Executing [s@macro-hangupcall:9] Hangup(SIP/1001-0014, ) in new stack == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/1001-0014' in macro 'hangupcall' == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/1001-0014' localhost*CLI Can someone assist me please. Thanks in advance. Regards, Malvin -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello
[asterisk-users] TDM400p susceptible to EMI?
I have a TDM400p with 3 fxs and 1 fxo daughter cards. It's in a mini-itx case with a 'right-angle' PCI riser card so the TDM400p is 'sandwiched' between the Atom D525 CPU and the 2.5 hard drive. I'm getting a bunch of clicks and pops on all ports. Has anybody had a similar experience? Did you find a solution? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM400p susceptible to EMI?
On Wed, Jul 13, 2011 at 2:51 PM, Steve Edwards asterisk@sedwards.com wrote: I have a TDM400p with 3 fxs and 1 fxo daughter cards. It's in a mini-itx case with a 'right-angle' PCI riser card so the TDM400p is 'sandwiched' between the Atom D525 CPU and the 2.5 hard drive. I'm getting a bunch of clicks and pops on all ports. Has anybody had a similar experience? Did you find a solution? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 How is it grounded? Silly I know but its possible. -- ~~~ Andrew lathama Latham lath...@gmail.com ~~~ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM400p susceptible to EMI?
- Original Message - I have a TDM400p with 3 fxs and 1 fxo daughter cards. It's in a mini-itx case with a 'right-angle' PCI riser card so the TDM400p is 'sandwiched' between the Atom D525 CPU and the 2.5 hard drive. I'm getting a bunch of clicks and pops on all ports. Has anybody had a similar experience? Did you find a solution? Many Mini-ITX cases have horrific power supplies to keep overall cost down. 'Cheap' doesn't begin to describe them. They *will* introduce audio issues, especially where you're using FXS modules. Also, it could be your system's ability to handle interrupts as that will cause clicks/pops as well. Third, try putting the card directly into the motherboard. PCI risers can be finicky, especially the 'ribbon style' units. The solid PCB units I've found are typically fine. --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM400p susceptible to EMI?
On Wed, Jul 13, 2011 at 2:51 PM, Steve Edwards asterisk@sedwards.com wrote: I have a TDM400p with 3 fxs and 1 fxo daughter cards. It's in a mini-itx case with a 'right-angle' PCI riser card so the TDM400p is 'sandwiched' between the Atom D525 CPU and the 2.5 hard drive. I'm getting a bunch of clicks and pops on all ports. Has anybody had a similar experience? Did you find a solution? On Wed, 13 Jul 2011, Andrew Latham wrote: How is it grounded? Silly I know but its possible. This box is using a picoPSU-80 80w DC-DC 'power supply' fed from an inline 'laptop brick.' I ran a separate lead from the chassis to the grounding plug on the same 'duplex' wall outlet. No joy. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM400p susceptible to EMI?
Sent from my Toshiba Satellite A106 computer -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards Sent: Wednesday, July 13, 2011 5:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] TDM400p susceptible to EMI? On Wed, Jul 13, 2011 at 2:51 PM, Steve Edwards asterisk@sedwards.com wrote: I have a TDM400p with 3 fxs and 1 fxo daughter cards. It's in a mini-itx case with a 'right-angle' PCI riser card so the TDM400p is 'sandwiched' between the Atom D525 CPU and the 2.5 hard drive. I'm getting a bunch of clicks and pops on all ports. Has anybody had a similar experience? Did you find a solution? On Wed, 13 Jul 2011, Andrew Latham wrote: How is it grounded? Silly I know but its possible. This box is using a picoPSU-80 80w DC-DC 'power supply' fed from an inline 'laptop brick.' I ran a separate lead from the chassis to the grounding plug on the same 'duplex' wall outlet. No joy. cat /proc/interrupts will tell you if the card is sharing IRQs with anything else. dahdi_tool should show you if there are any missed interrupts. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM400p susceptible to EMI?
- Original Message - On Wed, Jul 13, 2011 at 2:51 PM, Steve Edwards asterisk@sedwards.com wrote: I have a TDM400p with 3 fxs and 1 fxo daughter cards. It's in a mini-itx case with a 'right-angle' PCI riser card so the TDM400p is 'sandwiched' between the Atom D525 CPU and the 2.5 hard drive. I'm getting a bunch of clicks and pops on all ports. Has anybody had a similar experience? Did you find a solution? On Wed, 13 Jul 2011, Andrew Latham wrote: How is it grounded? Silly I know but its possible. This box is using a picoPSU-80 80w DC-DC 'power supply' fed from an inline 'laptop brick.' I ran a separate lead from the chassis to the grounding plug on the same 'duplex' wall outlet. No joy. picoPSU's are typically pretty good. I wouldn't suspect it in this case then unless your power supply is underpowered for the hardware's current draw. --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM400p susceptible to EMI?
On Wed, Jul 13, 2011 at 5:16 PM, Tim Nelson tnel...@rockbochs.com wrote: - Original Message - On Wed, Jul 13, 2011 at 2:51 PM, Steve Edwards asterisk@sedwards.com wrote: I have a TDM400p with 3 fxs and 1 fxo daughter cards. It's in a mini-itx case with a 'right-angle' PCI riser card so the TDM400p is 'sandwiched' between the Atom D525 CPU and the 2.5 hard drive. I'm getting a bunch of clicks and pops on all ports. Has anybody had a similar experience? Did you find a solution? On Wed, 13 Jul 2011, Andrew Latham wrote: How is it grounded? Silly I know but its possible. This box is using a picoPSU-80 80w DC-DC 'power supply' fed from an inline 'laptop brick.' I ran a separate lead from the chassis to the grounding plug on the same 'duplex' wall outlet. No joy. picoPSU's are typically pretty good. I wouldn't suspect it in this case then unless your power supply is underpowered for the hardware's current draw. --Tim I also use the pico-PSUs and have not had any issues. -- ~~~ Andrew lathama Latham lath...@gmail.com ~~~ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM400p susceptible to EMI?
On Wed, Jul 13, 2011 at 2:51 PM, Steve Edwards asterisk@sedwards.com wrote: I have a TDM400p with 3 fxs and 1 fxo daughter cards. It's in a mini-itx case with a 'right-angle' PCI riser card so the TDM400p is 'sandwiched' between the Atom D525 CPU and the 2.5 hard drive. I'm getting a bunch of clicks and pops on all ports. Has anybody had a similar experience? Did you find a solution? On Wed, 13 Jul 2011, Tim Nelson wrote: Many Mini-ITX cases have horrific power supplies to keep overall cost down. 'Cheap' doesn't begin to describe them. They *will* introduce audio issues, especially where you're using FXS modules. This box is using a picoPSU-80 80w DC-DC 'power supply' fed from an inline 'laptop brick.' Also, it could be your system's ability to handle interrupts as that will cause clicks/pops as well. Out of 300 samples from dahdi_test, there were 3 'outliers' -- 70.317%, 87.397%, 95.899% which is not comforting, but they don't correlate with the stream of noise. Third, try putting the card directly into the motherboard. PCI risers can be finicky, especially the 'ribbon style' units. The solid PCB units I've found are typically fine. This case will not accommodate that unfortunately. This is a rigid PCB unit. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM400p susceptible to EMI?
On Wed, 13 Jul 2011, Eric Wieling wrote: cat /proc/interrupts will tell you if the card is sharing IRQs with anything else. The card is on it's own on interrupt 66. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM400p susceptible to EMI?
- Original Message - Many Mini-ITX cases have horrific power supplies to keep overall cost down. 'Cheap' doesn't begin to describe them. They *will* introduce audio issues, especially where you're using FXS modules. This box is using a picoPSU-80 80w DC-DC 'power supply' fed from an inline 'laptop brick.' Also, it could be your system's ability to handle interrupts as that will cause clicks/pops as well. Out of 300 samples from dahdi_test, there were 3 'outliers' -- 70.317%, 87.397%, 95.899% which is not comforting, but they don't correlate with the stream of noise. Third, try putting the card directly into the motherboard. PCI risers can be finicky, especially the 'ribbon style' units. The solid PCB units I've found are typically fine. This case will not accommodate that unfortunately. This is a rigid PCB unit. Are you getting these audio artifacts on all channels, or specific ones? FXO vs FXS? Also, is the card a Digium TDM400P or a 'clone'? --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Chan_mobile
I am encountering problem recently with the chan_mobile that the bluetooth
connection between the asterisk and my Nokia E71 mobile phone frequently
connect and disconnect within seconds. As a result, I can't make any call
using Mobile/E71/{exten:2}.
Any suggested cause?
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Re: [asterisk-users] TDM400p susceptible to EMI?
On Wed, 2011-07-13 at 17:19 -0400, Andrew Latham wrote: On Wed, Jul 13, 2011 at 5:16 PM, Tim Nelson tnel...@rockbochs.com wrote: - Original Message - On Wed, Jul 13, 2011 at 2:51 PM, Steve Edwards asterisk@sedwards.com wrote: I have a TDM400p with 3 fxs and 1 fxo daughter cards. It's in a mini-itx case with a 'right-angle' PCI riser card so the TDM400p is 'sandwiched' between the Atom D525 CPU and the 2.5 hard drive. I'm getting a bunch of clicks and pops on all ports. Has anybody had a similar experience? Did you find a solution? On Wed, 13 Jul 2011, Andrew Latham wrote: How is it grounded? Silly I know but its possible. This box is using a picoPSU-80 80w DC-DC 'power supply' fed from an inline 'laptop brick.' I ran a separate lead from the chassis to the grounding plug on the same 'duplex' wall outlet. No joy. picoPSU's are typically pretty good. I wouldn't suspect it in this case then unless your power supply is underpowered for the hardware's current draw. --Tim I also use the pico-PSUs and have not had any issues. I do too - have dozens in the field. We use Rhino cards though, and no FXS (only FXO and T1). Also constrained by the case and use a horizontal riser, dual Atom. No hard drives though - 4G Sata flash drives. j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Extension wise dialplan
Hi all, I have n no. of extensions in my dialer. from 456 to 556 extensions. I was created 2 other extensions 667 and 668 I need to allow only STD calls to go from this extensions. These all extensions are same context . I need to define the STD dialplan for only this 2 extensions. how I can ? Best Regards, Mahesh Katta *BUZZ**WORKS* Business Services Private Limited BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI 201, Crystal Tower, 75 Gundavali Cross Lane, Andheri Kurla Road, Andheri (E) Mumbai 400069 GSM +91.97029.70779 | Phone +91.22.4229.2634 | Fax +91.22.4229.2634 Web http://www.buzzworks.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
