Re: [asterisk-users] help with dialplan
Here's the updated debug log. http:/www.computerworkx.net/client/Document.txt On 8/30/2010 2:55 PM, Paul Belanger wrote: On Mon, Aug 30, 2010 at 2:48 PM, Todd Reesetrees...@gmail.com wrote: Thanks for pointing out the misspelling. I've corrected that and still no luck. Create a new debug log with your recent changes, re-attach it the list. -- _ -- 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] help with dialplan
asterisk*CLI dialplan show 6789542...@remote There is no existence of 'remote' context Command 'dialplan show 6789542...@remote' failed. asterisk*CLI On 8/31/2010 9:58 AM, Paul Belanger wrote: dialplan show 6789542...@remote -- _ -- 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] help with dialplan
From extensions.conf
[remote]
include = from-internal
include = dialout1
include = dialout2
include = dialout3
include = intercom
exten = 150,1,Macro(oneline,${EXTERNPHONE0})
[dialout1]
include = from-internal
include = 411
include = remote
exten = 911,1,Goto(nineoneone,s,1)
exten = _1NXXNXX,n,Dial(SIP/v6787500600/${EXTEN},40,Ttr)
[dialout2]
include = from-internal
include = 411
include = remote
exten = 911,1,Goto(nineoneone,s,1)
exten = _1NXXNXX,n,Dial(SIP/voip.comACA/${EXTEN},40,Ttr)
[dialout3]
include = from-internal
include = 411
include = remote
exten = 911,1,Goto(nineoneone,s,1)
exten = _1NXXNXX,n,Dial(SIP/v6783747049/${EXTEN},40,Ttr)
On 8/31/2010 9:58 AM, Paul Belanger wrote:
On Tue, Aug 31, 2010 at 9:16 AM, Todd Reesetrees...@gmail.com wrote:
Here's the updated debug log.
[Aug 30 15:27:41] NOTICE[11568] chan_sip.c: Call from '150' to
extension '6789542133' rejected because extension not found in context
'remote'.
So, again, a context problem. You can confirm by entering:
*CLI dialplan show 6789542...@remote
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Re: [asterisk-users] help with dialplan
I had already check on this. Thanks for the info, though.
On 8/31/2010 10:36 AM, Danny Nicholas wrote:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Todd Reese
Subject: Re: [asterisk-users] help with dialplan
asterisk*CLI dialplan show 6789542...@remote
There is no existence of 'remote' context
Command 'dialplan show 6789542...@remote' failed.
asterisk*CLI
On 8/31/2010 9:58 AM, Paul Belanger wrote:
dialplan show 6789542...@remote
Ok. I'm a late joiner to this thread. Reading the original post I see
that you are trying to do an external SIP dial to 678-954-2133. These
questions:
1. Does the dial string need to be 1xxx instead of xxx (presumably local 10
digit dialing)? If yes, change
exten = _NXXNXX,n,Dial(SIP/v6781234567/${EXTEN},40,Ttr)
to
exten = _NXXNXX,n,Dial(SIP/v6781234567/1${EXTEN},40,Ttr)
2. voipdialACA and v6781234567 are registered trunks with credentials?
Hope this helps.
--
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Re: [asterisk-users] help with dialplan
Interesting things going on herel.
After your suggestions, Steve. I reran the dialplan show
16789542...@remote command with the below results.
Phone calls are geting the 404 error and the NOTICE on the console.
[Aug 31 11:06:46] NOTICE[11884]: chan_sip.c:20161 handle_request_invite:
Call from '150' to extension '16789542133' rejected because extension
not found in context 'remote'.
asterisk*CLI dialplan show 16789542...@remote
[ Included context 'dialout1' created by 'pbx_config' ]
'_1NXXNXX' = -2. Dial(SIP/v6787500600/${EXTEN},40,Ttr)
[pbx_config]
[ Included context 'dialout1' created by 'pbx_config' ]
'_1NXXNXX' = -2. Dial(SIP/v6787500600/${EXTEN},40,Ttr)
[pbx_config]
[ Included context 'dialout2' created by 'pbx_config' ]
'_1NXXNXX' = -2. Dial(SIP/voip.comACA/${EXTEN},40,Ttr)
[pbx_config]
[ Included context 'dialout3' created by 'pbx_config' ]
'_1NXXNXX' = -2. Dial(SIP/v6783747049/${EXTEN},40,Ttr)
[pbx_config]
[ Included context 'dialout1' created by 'pbx_config' ]
'_1NXXNXX' = -2. Dial(SIP/v6787500600/${EXTEN},40,Ttr)
[pbx_config]
[ Included context 'dialout2' created by 'pbx_config' ]
'_1NXXNXX' = -2. Dial(SIP/voip.comACA/${EXTEN},40,Ttr)
[pbx_config]
[ Included context 'dialout3' created by 'pbx_config' ]
'_1NXXNXX' = -2. Dial(SIP/v6783747049/${EXTEN},40,Ttr)
[pbx_config]
-= 7 extensions (7 priorities) in 7 contexts. =-
[Aug 31 11:03:38] WARNING[11903]: pbx.c:5741 show_dialplan_helper:
Avoiding circular include of from-internal within remote
On 8/31/2010 10:49 AM, Steve Murphy wrote:
Todd--
There is probably some nifty anti-infinite-recursion code in the
extensions.conf parser,
to keep asterisk from going into infinite loops trying to descend into
the right context.
In your dialplan, [remote] includes dialout1, dialout2, dialout3, and
each of those
include remote.
Straighten out that mess and maybe things might work. Just a guess,
but worth a try!
murf
On Tue, Aug 31, 2010 at 8:25 AM, Todd Reese trees...@gmail.com
mailto:trees...@gmail.com wrote:
From extensions.conf
[remote]
include = from-internal
include = dialout1
include = dialout2
include = dialout3
include = intercom
exten = 150,1,Macro(oneline,${EXTERNPHONE0})
[dialout1]
include = from-internal
include = 411
include = remote
exten = 911,1,Goto(nineoneone,s,1)
exten = _1NXXNXX,n,Dial(SIP/v6787500600/${EXTEN},40,Ttr)
[dialout2]
include = from-internal
include = 411
include = remote
exten = 911,1,Goto(nineoneone,s,1)
exten = _1NXXNXX,n,Dial(SIP/voip.comACA/${EXTEN},40,Ttr)
[dialout3]
include = from-internal
include = 411
include = remote
exten = 911,1,Goto(nineoneone,s,1)
exten = _1NXXNXX,n,Dial(SIP/v6783747049/${EXTEN},40,Ttr)
On 8/31/2010 9:58 AM, Paul Belanger wrote:
On Tue, Aug 31, 2010 at 9:16 AM, Todd Reesetrees...@gmail.com
mailto:trees...@gmail.com wrote:
Here's the updated debug log.
[Aug 30 15:27:41] NOTICE[11568] chan_sip.c: Call from '150' to
extension '6789542133' rejected because extension not found in
context
'remote'.
So, again, a context problem. You can confirm by entering:
*CLI dialplan show 6789542...@remote
--
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--
Steve Murphy
ParseTree Corp
--
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[asterisk-users] help with dialplan
Hi all,
I've been have problems with getting this system on line and would like
to acquire some help with the extensions.conf.
My current problem is that the phones won't dialout.on the VOIP lines
listed as dialout1, dialout2, dialout3. This version of asterisk is
1.6.2.11. Below is the extensions.conf file.
[globals]
QPHONE0=SIP/10
QPHONE1=SIP/11
QPHONE2=SIP/12
QPHONE3=SIP/13
QPHONE4=SIP/14
QPHONE5=SIP/15
QPHONE6=SIP/16
QPHONE7=SIP/17
ACAPHONE0=SIP/20
ACAPHONE1=SIP/21
ACAPHONE2=SIP/22
ACAPHONE3=SIP/23
ACAPHONE4=SIP/24
ACAPHONE5=SIP/25
ACAPHONE6=SIP/26
ACAPHONE7=SIP/27
GMNETPHONE0=SIP/30
GMNETPHONE1=SIP/31
GMNETPHONE2=SIP/32
GMNETPHONE3=SIP/33
GMNETPHONE4=SIP/34
GMNETPHONE5=SIP/35
GMNETPHONE6=SIP/36
GMNETPHONE7=SIP/37
EXTERNPHONE0=SIP/150
CPHONE1=SIP/1678000
CPHONE2=SIP/177
EMERGENCY=0
EMERGENCY_TRUNK=DAHDI/G1
; Change this for production use:
EMERGENCY_NUM=6789542133
[from-pstn]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming1,s,1)
[from-pstn1]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming1,s,1)
[from-pstn2]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming1,s,1)
[from-pstn3]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming1,s,1)
[from-pstn4]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming3,s,1)
[from-pstn5]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming2,s,1)
[from-pstn6]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming1,s,1)
[from-pstn7]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming1,s,1)
[from-pstn8]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming1,s,1)
[incoming1]
include = from-internal
include = parkedcalls
exten = s,1,Answer
exten = s,n,Wait(1)
exten = s,n,Set(CHANNEL(musicclass)=QCI)
exten = s,n,Set(TIMEOUT(digit)=5)
exten = s,n,Set(TIMEOUT(response)=10)
exten = s,n,Background(thank-you-for-calling)
exten =
s,n,Dial(${QPHONE0}${QPHONE1}${QPHONE2}${QPHONE3}${QPHONE4}${QPHONE5}${QPHONE6}${QPHONE7},40,Ttr)
exten = s,n,Hangup
[incoming2]
include = from-internal
include = parkedcalls
exten = s,1,Answer
exten = s,n,Wait(1)
exten = s,n,Set(CHANNEL(musicclass)=QCI)
exten = s,n,Set(TIMEOUT(digit)=5)
exten = s,n,Set(TIMEOUT(response)=10)
exten = s,n,Background(thank-you-for-calling)
exten =
s,n,Dial(${ACAPHONE0}${ACAPHONE1}${ACAPHONE2}${ACAPHONE3}${ACAPHONE4}${ACAPHONE5}${ACAPHONE6}${ACAPHONE7},40,TTr)
exten = s,n,Hangup
[incoming3]
include = from-internal
include = parkedcalls
exten = s,1,Answer
exten = s,n,Wait(1)
exten = s,n,Set(CHANNEL(musicclass)=QCI)
exten = s,n,Set(TIMEOUT(digit)=5)
exten = s,n,Set(TIMEOUT(response)=10)
exten = s,n,Background(thank-you-for-calling)
exten = s,n,GotoIfTime(17:01-09:00|*|*|*?closed,s,1)
exten =
s,n,Dial(${GMNETPHONE0}${GMNETPHONE1}${GMNETPHONE2}${GMNETPHONE3}${GMNETPHONE4}${GMNETPHONE5}${GMNETPHONE6}${GMNETPHONE7},40,Ttr)
exten = s,n,Hangup
[from-interal]
include = dialout1
include = dialout2
include = dialout3
include = parkedcalls
include = intercom
exten = 10,1,Macro(oneline,${QPHONE0})
exten = 11,1,Macro(oneline,${QPHONE1})
exten = 12,1,Macro(oneline,${QPHONE2})
exten = 13,1,Macro(oneline,${QPHONE3})
exten = 14,1,Macro(oneline,${QPHONE4})
exten = 15,1,Macro(oneline,${QPHONE5})
exten = 16,1,Macro(oneline,${QPHONE6})
exten = 17,1,Macro(oneline,${QPHONE7})
exten = 20,1,Macro(oneline,${ACAPHONE0})
exten = 21,1,Macro(oneline,${ACAPHONE1})
exten = 22,1,Macro(oneline,${ACAPHONE2})
exten = 23,1,Macro(oneline,${ACAPHONE3})
exten = 24,1,Macro(oneline,${ACAPHONE4})
exten = 25,1,Macro(oneline,${ACAPHONE5})
exten = 26,1,Macro(oneline,${ACAPHONE6})
exten = 27,1,Macro(oneline,${ACAPHONE7})
exten = 30,1,Macro(oneline,${GMNETPHONE0})
exten = 31,1,Macro(oneline,${GMNETPHONE1})
exten = 32,1,Macro(oneline,${GMNETPHONE2})
exten = 33,1,Macro(oneline,${GMNETPHONE3})
exten = 34,1,Macro(oneline,${GMNETPHONE4})
exten = 35,1,Macro(oneline,${GMNETPHONE5})
exten = 36,1,Macro(oneline,${GMNETPHONE6})
exten = 37,1,Macro(oneline,${GMNETPHONE7})
exten = 40,1,Macro(oneline,${QPHONE0})
exten = 41,1,Macro(oneline,${QPHONE1})
exten = 42,1,Macro(oneline,${QPHONE2})
exten = 43,1,Macro(oneline,${QPHONE3})
exten = 44,1,Macro(oneline,${QPHONE4})
exten = 45,1,Macro(oneline,${QPHONE5})
exten = 46,1,Macro(oneline,${QPHONE6})
exten = 47,1,Macro(oneline,${QPHONE7})
exten = 150,1,Macro(oneline,${EXTERNPHONE0})
[macro-oneline]
exten = s,1,Set(CHANNEL(musicclass)=default)
exten = s,n,Dial(${ARG1},20,Ttr)
exten = s,n,Voicemail(${MACRO_EXTEN})
exten = s,n,Hangup
exten = s,102,Voicemail(${MACRO_EXTEN})
exten = s,103,Hangup
[dialout1]
include = from-internal
include = 411
exten =
Re: [asterisk-users] help with dialplan
Here is the sip.conf portion for extension 150
[150]
deny=0.0.0.0/0.0.0.0
type=friend
secret=1234567890
qualify=yes
port=5060
pickupgroup=
permit=0.0.0.0/0.0.0.0
nat=yes
host=dynamic
dtmfmode=rfc2833
dial=SIP/150
context=from-trunk
canreinvite=no
callgroup=
callerid=device 150
accountcode=
call-limit=50
On 8/30/2010 10:37 AM, Bryant Zimmerman wrote:
Todd
How do you have the context in the phones sip configs set?
Bryant
*
From*: Todd Reese trees...@gmail.com mailto:trees...@gmail.com
Hi all,
I've been have problems with getting this system on line and would like
to acquire some help with the extensions.conf.
My current problem is that the phones won't dialout.on the VOIP lines
listed as dialout1, dialout2, dialout3. This version of asterisk is
1.6.2.11. Below is the extensions.conf file.
[globals]
QPHONE0=SIP/10
QPHONE1=SIP/11
QPHONE2=SIP/12
QPHONE3=SIP/13
QPHONE4=SIP/14
QPHONE5=SIP/15
QPHONE6=SIP/16
QPHONE7=SIP/17
ACAPHONE0=SIP/20
ACAPHONE1=SIP/21
ACAPHONE2=SIP/22
ACAPHONE3=SIP/23
ACAPHONE4=SIP/24
ACAPHONE5=SIP/25
ACAPHONE6=SIP/26
ACAPHONE7=SIP/27
GMNETPHONE0=SIP/30
GMNETPHONE1=SIP/31
GMNETPHONE2=SIP/32
GMNETPHONE3=SIP/33
GMNETPHONE4=SIP/34
GMNETPHONE5=SIP/35
GMNETPHONE6=SIP/36
GMNETPHONE7=SIP/37
EXTERNPHONE0=SIP/150
CPHONE1=SIP/1678000
CPHONE2=SIP/177
EMERGENCY=0
EMERGENCY_TRUNK=DAHDI/G1
; Change this for production use:
EMERGENCY_NUM=6789542133
[from-pstn]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming1,s,1)
[from-pstn1]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming1,s,1)
[from-pstn2]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming1,s,1)
[from-pstn3]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming1,s,1)
[from-pstn4]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming3,s,1)
[from-pstn5]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming2,s,1)
[from-pstn6]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming1,s,1)
[from-pstn7]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming1,s,1)
[from-pstn8]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming1,s,1)
[incoming1]
include = from-internal
include = parkedcalls
exten = s,1,Answer
exten = s,n,Wait(1)
exten = s,n,Set(CHANNEL(musicclass)=QCI)
exten = s,n,Set(TIMEOUT(digit)=5)
exten = s,n,Set(TIMEOUT(response)=10)
exten = s,n,Background(thank-you-for-calling)
exten =
s,n,Dial(${QPHONE0}${QPHONE1}${QPHONE2}${QPHONE3}${QPHONE4}${QPHONE5}${QPHONE6}${QPHONE7},40,Ttr)
exten = s,n,Hangup
[incoming2]
include = from-internal
include = parkedcalls
exten = s,1,Answer
exten = s,n,Wait(1)
exten = s,n,Set(CHANNEL(musicclass)=QCI)
exten = s,n,Set(TIMEOUT(digit)=5)
exten = s,n,Set(TIMEOUT(response)=10)
exten = s,n,Background(thank-you-for-calling)
exten =
s,n,Dial(${ACAPHONE0}${ACAPHONE1}${ACAPHONE2}${ACAPHONE3}${ACAPHONE4}${ACAPHONE5}${ACAPHONE6}${ACAPHONE7},40,TTr)
exten = s,n,Hangup
[incoming3]
include = from-internal
include = parkedcalls
exten = s,1,Answer
exten = s,n,Wait(1)
exten = s,n,Set(CHANNEL(musicclass)=QCI)
exten = s,n,Set(TIMEOUT(digit)=5)
exten = s,n,Set(TIMEOUT(response)=10)
exten = s,n,Background(thank-you-for-calling)
exten = s,n,GotoIfTime(17:01-09:00|*|*|*?closed,s,1)
exten =
s,n,Dial(${GMNETPHONE0}${GMNETPHONE1}${GMNETPHONE2}${GMNETPHONE3}${GMNETPHONE4}${GMNETPHONE5}${GMNETPHONE6}${GMNETPHONE7},40,Ttr)
exten = s,n,Hangup
[from-interal]
include = dialout1
include = dialout2
include = dialout3
include = parkedcalls
include = intercom
exten = 10,1,Macro(oneline,${QPHONE0})
exten = 11,1,Macro(oneline,${QPHONE1})
exten = 12,1,Macro(oneline,${QPHONE2})
exten = 13,1,Macro(oneline,${QPHONE3})
exten = 14,1,Macro(oneline,${QPHONE4})
exten = 15,1,Macro(oneline,${QPHONE5})
exten = 16,1,Macro(oneline,${QPHONE6})
exten = 17,1,Macro(oneline,${QPHONE7})
exten = 20,1,Macro(oneline,${ACAPHONE0})
exten = 21,1,Macro(oneline,${ACAPHONE1})
exten = 22,1,Macro(oneline,${ACAPHONE2})
exten = 23,1,Macro(oneline,${ACAPHONE3})
exten = 24,1,Macro(oneline,${ACAPHONE4})
exten = 25,1,Macro(oneline,${ACAPHONE5})
exten = 26,1,Macro(oneline,${ACAPHONE6})
exten = 27,1,Macro(oneline,${ACAPHONE7})
exten = 30,1,Macro(oneline,${GMNETPHONE0})
exten = 31,1,Macro(oneline,${GMNETPHONE1})
exten = 32,1,Macro(oneline,${GMNETPHONE2})
exten = 33,1,Macro(oneline,${GMNETPHONE3})
exten = 34,1,Macro(oneline,${GMNETPHONE4})
exten = 35,1,Macro(oneline,${GMNETPHONE5})
exten = 36,1,Macro(oneline,${GMNETPHONE6})
exten = 37,1,Macro(oneline,${GMNETPHONE7})
exten = 40,1,Macro(oneline,${QPHONE0})
exten = 41,1,Macro(oneline,${QPHONE1})
exten = 42,1,Macro(oneline,${QPHONE2})
exten = 43,1,Macro(oneline,${QPHONE3})
exten = 44,1
Re: [asterisk-users] help with dialplan
Here's a debug for extension 150 [Aug 30 11:34:53] VERBOSE[2099] config.c: == Parsing '/etc/asterisk/logger.conf': [Aug 30 11:34:53] DEBUG[2099] config.c: Parsing /etc/asterisk/logger.conf [Aug 30 11:34:53] VERBOSE[2099] config.c: == Found [Aug 30 11:34:53] VERBOSE[2099] logger.c: Asterisk Event Logger restarted [Aug 30 11:34:53] VERBOSE[2099] logger.c: Asterisk Queue Logger restarted [Aug 30 11:34:53] DEBUG[2079] chan_sip.c: Header 0 [ 38]: OPTIONS sip:76.122.117.31:5060 SIP/2.0 [Aug 30 11:34:53] DEBUG[2079] chan_sip.c: Header 1 [ 44]: Via: SIP/2.0/UDP 64.34.245.174:5060;branch=0 [Aug 30 11:34:53] DEBUG[2079] chan_sip.c: Header 2 [ 38]: From: sip:pin...@voip.com;tag=7c9c6206 [Aug 30 11:34:53] DEBUG[2079] chan_sip.c: Header 3 [ 26]: To: sip:76.122.117.31:5060 [Aug 30 11:34:53] DEBUG[2079] chan_sip.c: Header 4 [ 47]: Call-ID: 89c833e4-9c8f2516-5bb...@64.34.245.174 [Aug 30 11:34:53] DEBUG[2079] chan_sip.c: Header 5 [ 15]: CSeq: 1 OPTIONS [Aug 30 11:34:53] DEBUG[2079] chan_sip.c: Header 6 [ 17]: Content-Length: 0 [Aug 30 11:34:53] DEBUG[2079] chan_sip.c: Header 7 [ 0]: [Aug 30 11:34:53] DEBUG[2079] acl.c: Found IP address for this socket [Aug 30 11:34:53] DEBUG[2079] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 10.0.1.102:5060 [Aug 30 11:34:53] DEBUG[2079] chan_sip.c: Allocating new SIP dialog for 89c833e4-9c8f2516-5bb...@64.34.245.174 - OPTIONS (No RTP) [Aug 30 11:34:53] DEBUG[2079] chan_sip.c: Received OPTIONS (3) - Command in SIP OPTIONS [Aug 30 11:34:53] DEBUG[2079] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 64.34.245.174:5060 [Aug 30 11:34:53] DEBUG[2079] chan_sip.c: SIP message could not be handled, bad request: 89c833e4-9c8f2516-5bb...@64.34.245.174 [Aug 30 11:34:53] DEBUG[2079] chan_sip.c: Header 0 [ 38]: OPTIONS sip:76.122.117.31:5060 SIP/2.0 [Aug 30 11:34:53] DEBUG[2079] chan_sip.c: Header 1 [ 44]: Via: SIP/2.0/UDP 64.34.245.174:5060;branch=0 [Aug 30 11:34:53] DEBUG[2079] chan_sip.c: Header 2 [ 38]: From: sip:pin...@voip.com;tag=1f9c6206 [Aug 30 11:34:53] DEBUG[2079] chan_sip.c: Header 3 [ 26]: To: sip:76.122.117.31:5060 [Aug 30 11:34:53] DEBUG[2079] chan_sip.c: Header 4 [ 47]: Call-ID: 89c833e4-3f8f2516-5bb...@64.34.245.174 [Aug 30 11:34:53] DEBUG[2079] chan_sip.c: Header 5 [ 15]: CSeq: 1 OPTIONS [Aug 30 11:34:53] DEBUG[2079] chan_sip.c: Header 6 [ 17]: Content-Length: 0 [Aug 30 11:34:53] DEBUG[2079] chan_sip.c: Header 7 [ 0]: [Aug 30 11:34:53] DEBUG[2079] acl.c: Found IP address for this socket [Aug 30 11:34:53] DEBUG[2079] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 10.0.1.102:5060 [Aug 30 11:34:53] DEBUG[2079] chan_sip.c: Allocating new SIP dialog for 89c833e4-3f8f2516-5bb...@64.34.245.174 - OPTIONS (No RTP) [Aug 30 11:34:53] DEBUG[2079] chan_sip.c: Received OPTIONS (3) - Command in SIP OPTIONS [Aug 30 11:34:53] DEBUG[2079] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 64.34.245.174:5060 [Aug 30 11:34:53] DEBUG[2079] chan_sip.c: SIP message could not be handled, bad request: 89c833e4-3f8f2516-5bb...@64.34.245.174 [Aug 30 11:34:54] DEBUG[2079] chan_sip.c: Auto destroying SIP dialog '89c833e4-806e2516-79b...@64.34.245.174' [Aug 30 11:34:54] DEBUG[2079] chan_sip.c: Destroying SIP dialog 89c833e4-806e2516-79b...@64.34.245.174 [Aug 30 11:34:54] DEBUG[2079] chan_sip.c: Auto destroying SIP dialog '89c833e4-236e2516-79b...@64.34.245.174' [Aug 30 11:34:54] DEBUG[2079] chan_sip.c: Destroying SIP dialog 89c833e4-236e2516-79b...@64.34.245.174 [Aug 30 11:34:54] VERBOSE[2079] chan_sip.c: --- SIP read from UDP:97.80.176.231:5060 --- - [Aug 30 11:34:54] DEBUG[2079] chan_sip.c: Header 0 [ 0]: [Aug 30 11:34:54] DEBUG[2079] chan_sip.c:Body 0 [ 0]: [Aug 30 11:34:54] VERBOSE[2079] chan_sip.c: --- SIP read from UDP:97.80.176.231:5060 --- INVITE sip:6789542...@qci.homeip.net SIP/2.0 Via: SIP/2.0/UDP 10.11.17.24:5060;branch=z9hG4bK82d40955a28df1af From: ATAP sip:1...@qci.homeip.net;tag=ee0cedf5f71d40f9 To: sip:6789542...@qci.homeip.net Contact: sip:1...@10.11.17.24:5060;transport=udp Supported: replaces, timer, path P-Early-Media: Supported Call-ID: 62f35b2ee0ada...@10.11.17.24 CSeq: 21395 INVITE User-Agent: Grandstream GXP2000 1.2.3.5 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Content-Length: 345 v=0 o=150 8000 8000 IN IP4 10.11.17.24 s=SIP Call c=IN IP4 10.11.17.24 t=0 0 m=audio 5050 RTP/AVP 0 8 4 18 2 97 9 3 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=20 a=rtpmap:9 G722/8000 a=rtpmap:3 GSM/8000 a=ptime:20 - [Aug 30 11:34:54] DEBUG[2079] chan_sip.c: Header 0 [ 44]: INVITE sip:6789542...@qci.homeip.net SIP/2.0 [Aug 30 11:34:54] DEBUG[2079] chan_sip.c: Header 1 [ 64]: Via: SIP/2.0/UDP 10.11.17.24:5060;branch=z9hG4bK82d40955a28df1af
Re: [asterisk-users] help with dialplan
Unfortunately, that didn't work. The phone is still giving me a 404
error.
I have my own system that is 1.6.2.7 with Grandstream phones that works
fine. Using it as a guide, I built this server for a client which also
has Grandstream phones.
Last week, it dialed out fine. Since the weekend, no dialing at all.
On 8/30/2010 11:42 AM, Bryant Zimmerman wrote:
Todd
Your context must be set to where you want your extension to start
each time it dials out. Without getting into your dialplan code too
much try changing the context to point to dialout1
context=dialout1
If dialout1 is working you should be able to dial.
The best way to handle this is to create a context that when you dial
from your phones it decieds if you have dialed an extension or an
external number and then routes the call correclty. This way you can
pickup an extension and dial either and get the desired results.
Bryant
*From*: Todd Reese trees...@gmail.com
*Sent*: Monday, August 30, 2010 11:20 AM
*To*: asterisk-users@lists.digium.com
*Subject*: Re: [asterisk-users] help with dialplan
Here is the sip.conf portion for extension 150
[150]
deny=0.0.0.0/0.0.0.0
type=friend
secret=1234567890
qualify=yes
port=5060
pickupgroup=
permit=0.0.0.0/0.0.0.0
nat=yes
host=dynamic
dtmfmode=rfc2833
dial=SIP/150
context=from-trunk
canreinvite=no
callgroup=
callerid=device 150
accountcode=
call-limit=50
On 8/30/2010 10:37 AM, Bryant Zimmerman wrote:
Todd
How do you have the context in the phones sip configs set?
Bryant
*
From*: Todd Reese trees...@gmail.com mailto:trees...@gmail.com
Hi all,
I've been have problems with getting this system on line and would like
to acquire some help with the extensions.conf.
My current problem is that the phones won't dialout.on the VOIP lines
listed as dialout1, dialout2, dialout3. This version of asterisk is
1.6.2.11. Below is the extensions.conf file.
[globals]
QPHONE0=SIP/10
QPHONE1=SIP/11
QPHONE2=SIP/12
QPHONE3=SIP/13
QPHONE4=SIP/14
QPHONE5=SIP/15
QPHONE6=SIP/16
QPHONE7=SIP/17
ACAPHONE0=SIP/20
ACAPHONE1=SIP/21
ACAPHONE2=SIP/22
ACAPHONE3=SIP/23
ACAPHONE4=SIP/24
ACAPHONE5=SIP/25
ACAPHONE6=SIP/26
ACAPHONE7=SIP/27
GMNETPHONE0=SIP/30
GMNETPHONE1=SIP/31
GMNETPHONE2=SIP/32
GMNETPHONE3=SIP/33
GMNETPHONE4=SIP/34
GMNETPHONE5=SIP/35
GMNETPHONE6=SIP/36
GMNETPHONE7=SIP/37
EXTERNPHONE0=SIP/150
CPHONE1=SIP/1678000
CPHONE2=SIP/177
EMERGENCY=0
EMERGENCY_TRUNK=DAHDI/G1
; Change this for production use:
EMERGENCY_NUM=6789542133
[from-pstn]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming1,s,1)
[from-pstn1]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming1,s,1)
[from-pstn2]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming1,s,1)
[from-pstn3]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming1,s,1)
[from-pstn4]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming3,s,1)
[from-pstn5]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming2,s,1)
[from-pstn6]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming1,s,1)
[from-pstn7]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming1,s,1)
[from-pstn8]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming1,s,1)
[incoming1]
include = from-internal
include = parkedcalls
exten = s,1,Answer
exten = s,n,Wait(1)
exten = s,n,Set(CHANNEL(musicclass)=QCI)
exten = s,n,Set(TIMEOUT(digit)=5)
exten = s,n,Set(TIMEOUT(response)=10)
exten = s,n,Background(thank-you-for-calling)
exten =
s,n,Dial(${QPHONE0}${QPHONE1}${QPHONE2}${QPHONE3}${QPHONE4}${QPHONE5}${QPHONE6}${QPHONE7},40,Ttr)
exten = s,n,Hangup
[incoming2]
include = from-internal
include = parkedcalls
exten = s,1,Answer
exten = s,n,Wait(1)
exten = s,n,Set(CHANNEL(musicclass)=QCI)
exten = s,n,Set(TIMEOUT(digit)=5)
exten = s,n,Set(TIMEOUT(response)=10)
exten = s,n,Background(thank-you-for-calling)
exten =
s,n,Dial(${ACAPHONE0}${ACAPHONE1}${ACAPHONE2}${ACAPHONE3}${ACAPHONE4}${ACAPHONE5}${ACAPHONE6}${ACAPHONE7},40,TTr)
exten = s,n,Hangup
[incoming3]
include = from-internal
include = parkedcalls
exten = s,1,Answer
exten = s,n,Wait(1)
exten = s,n,Set(CHANNEL(musicclass)=QCI)
exten = s,n,Set(TIMEOUT(digit)=5)
exten = s,n,Set(TIMEOUT(response)=10)
exten = s,n,Background(thank-you-for-calling)
exten = s,n,GotoIfTime(17:01-09:00|*|*|*?closed,s,1)
exten =
s,n,Dial(${GMNETPHONE0}${GMNETPHONE1}${GMNETPHONE2}${GMNETPHONE3}${GMNETPHONE4}${GMNETPHONE5}${GMNETPHONE6}${GMNETPHONE7},40,Ttr)
exten = s,n,Hangup
[from-interal]
include = dialout1
include = dialout2
include = dialout3
include = parkedcalls
include
Re: [asterisk-users] help with dialplan
I actually found that one and corrected it. I have replaced the context with the from-internal, remote, and dialout1. Each has produced the same results of a 404 error. On 8/30/2010 2:10 PM, Paul Belanger wrote: On Mon, Aug 30, 2010 at 11:42 AM, Todd Reesetrees...@gmail.com wrote: Here's a debug for extension 150 In the future, simply attach your debug log to your email. Here is your problem: [Aug 30 11:34:55] NOTICE[2079] chan_sip.c: Call from '150' to extension '6789542133' rejected because extension not found in context 'extensions.conf'. -- _ -- 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] help with dialplan
Thanks for pointing out the misspelling. I've corrected that and
still no luck.
On 8/30/2010 2:33 PM, Alex Bell wrote:
possibly check you spelling: [from-interal] - [dialout1]
include = from-internal
??
On Mon, Aug 30, 2010 at 10:14 AM, Todd Reese trees...@gmail.com
mailto:trees...@gmail.com wrote:
Hi all,
I've been have problems with getting this system on line and would
like
to acquire some help with the extensions.conf.
My current problem is that the phones won't dialout.on the VOIP lines
listed as dialout1, dialout2, dialout3. This version of asterisk is
1.6.2.11. Below is the extensions.conf file.
[globals]
QPHONE0=SIP/10
QPHONE1=SIP/11
QPHONE2=SIP/12
QPHONE3=SIP/13
QPHONE4=SIP/14
QPHONE5=SIP/15
QPHONE6=SIP/16
QPHONE7=SIP/17
ACAPHONE0=SIP/20
ACAPHONE1=SIP/21
ACAPHONE2=SIP/22
ACAPHONE3=SIP/23
ACAPHONE4=SIP/24
ACAPHONE5=SIP/25
ACAPHONE6=SIP/26
ACAPHONE7=SIP/27
GMNETPHONE0=SIP/30
GMNETPHONE1=SIP/31
GMNETPHONE2=SIP/32
GMNETPHONE3=SIP/33
GMNETPHONE4=SIP/34
GMNETPHONE5=SIP/35
GMNETPHONE6=SIP/36
GMNETPHONE7=SIP/37
EXTERNPHONE0=SIP/150
CPHONE1=SIP/1678000
CPHONE2=SIP/177
EMERGENCY=0
EMERGENCY_TRUNK=DAHDI/G1
; Change this for production use:
EMERGENCY_NUM=6789542133
[from-pstn]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming1,s,1)
[from-pstn1]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming1,s,1)
[from-pstn2]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming1,s,1)
[from-pstn3]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming1,s,1)
[from-pstn4]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming3,s,1)
[from-pstn5]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming2,s,1)
[from-pstn6]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming1,s,1)
[from-pstn7]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming1,s,1)
[from-pstn8]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming1,s,1)
[incoming1]
include = from-internal
include = parkedcalls
exten = s,1,Answer
exten = s,n,Wait(1)
exten = s,n,Set(CHANNEL(musicclass)=QCI)
exten = s,n,Set(TIMEOUT(digit)=5)
exten = s,n,Set(TIMEOUT(response)=10)
exten = s,n,Background(thank-you-for-calling)
exten =
s,n,Dial(${QPHONE0}${QPHONE1}${QPHONE2}${QPHONE3}${QPHONE4}${QPHONE5}${QPHONE6}${QPHONE7},40,Ttr)
exten = s,n,Hangup
[incoming2]
include = from-internal
include = parkedcalls
exten = s,1,Answer
exten = s,n,Wait(1)
exten = s,n,Set(CHANNEL(musicclass)=QCI)
exten = s,n,Set(TIMEOUT(digit)=5)
exten = s,n,Set(TIMEOUT(response)=10)
exten = s,n,Background(thank-you-for-calling)
exten =
s,n,Dial(${ACAPHONE0}${ACAPHONE1}${ACAPHONE2}${ACAPHONE3}${ACAPHONE4}${ACAPHONE5}${ACAPHONE6}${ACAPHONE7},40,TTr)
exten = s,n,Hangup
[incoming3]
include = from-internal
include = parkedcalls
exten = s,1,Answer
exten = s,n,Wait(1)
exten = s,n,Set(CHANNEL(musicclass)=QCI)
exten = s,n,Set(TIMEOUT(digit)=5)
exten = s,n,Set(TIMEOUT(response)=10)
exten = s,n,Background(thank-you-for-calling)
exten = s,n,GotoIfTime(17:01-09:00|*|*|*?closed,s,1)
exten =
s,n,Dial(${GMNETPHONE0}${GMNETPHONE1}${GMNETPHONE2}${GMNETPHONE3}${GMNETPHONE4}${GMNETPHONE5}${GMNETPHONE6}${GMNETPHONE7},40,Ttr)
exten = s,n,Hangup
[from-interal]
include = dialout1
include = dialout2
include = dialout3
include = parkedcalls
include = intercom
exten = 10,1,Macro(oneline,${QPHONE0})
exten = 11,1,Macro(oneline,${QPHONE1})
exten = 12,1,Macro(oneline,${QPHONE2})
exten = 13,1,Macro(oneline,${QPHONE3})
exten = 14,1,Macro(oneline,${QPHONE4})
exten = 15,1,Macro(oneline,${QPHONE5})
exten = 16,1,Macro(oneline,${QPHONE6})
exten = 17,1,Macro(oneline,${QPHONE7})
exten = 20,1,Macro(oneline,${ACAPHONE0})
exten = 21,1,Macro(oneline,${ACAPHONE1})
exten = 22,1,Macro(oneline,${ACAPHONE2})
exten = 23,1,Macro(oneline,${ACAPHONE3})
exten = 24,1,Macro(oneline,${ACAPHONE4})
exten = 25,1,Macro(oneline,${ACAPHONE5})
exten = 26,1,Macro(oneline,${ACAPHONE6})
exten = 27,1,Macro(oneline,${ACAPHONE7})
exten = 30,1,Macro(oneline,${GMNETPHONE0})
exten = 31,1,Macro(oneline,${GMNETPHONE1})
exten = 32,1,Macro(oneline,${GMNETPHONE2})
exten = 33,1
[asterisk-users] GXP-2000 transfer hold problem
Hi all, I'm working on a system with 4 Grandstream GP-200 Phones and the base Asterisk install. I have added a 5 phone which is remote to the client and located in my office. I can't get the phone to transfer a call or put a call on hold. This applies to all the phones at the location. I have been looking over configs and I'm at a loss right now. Any help in pointing this out would be greatly appreciated. -- _ -- 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] Dahdi install gone wrong
Hi All, I've got a project installing a Digium TDM800P card with 8 FXO's in an Asterisk box. The computer is running Slackware 13.1 and I've installed the current Dahdi and Asterisk 1.6.2.11. I've installed several boxes that are pure VOIP but, I haven't installed a Dahdi interface and I'm stumped. I've got it to the point of Dahdi seeing the card and Asterisk recognizing dahdi but, I can't see any channels for the calls to come in on. I've had to borrow files from an old config of Trixbox (the machine was underpowered) to get to the point where I am in my setup. I would like to inquire some help from the group to get me up and receiving calls on the card. Regards, Todd Reese Include: chan_dahdi.conf== ; Configuration file [trunkgroups] [channels] language=en context=from-zaptel signalling=fxs_ks rxwink=300 ; Atlas seems to use long (250ms) winks ; ; Whether or not to do distinctive ring detection on FXO lines ; ;usedistinctiveringdetection=yes usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no ;echotraining=800 rxgain=0.0 txgain=0.0 group=0 callgroup=1 pickupgroup=1 immediate=no ;faxdetect=both faxdetect=incoming ;faxdetect=outgoing ;faxdetect=no ;Include setup-pstn configs #include dahdi-channels.conf group=1 ;Include PBXconfig configs #include chan_dahdi_additional.conf dahdi-channels.conf= ; Autogenerated by /usr/sbin/dahdi_genconf on Wed Aug 18 20:25:02 2010 ; If you edit this file and execute /usr/sbin/dahdi_genconf again, ; your manual changes will be LOST. ; 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/chan_dahdi.conf that will include the global settings ; ; Span 1: WCTDM/0 Wildcard TDM800P Board 1 (MASTER) ;;; line=1 WCTDM/0/0 FXSKS (SWEC: MG2) signalling=fxs_ks callerid=asreceived group=0 context=from-pstn channel = 1 callerid= group= context=default ;;; line=2 WCTDM/0/1 FXSKS (SWEC: MG2) signalling=fxs_ks callerid=asreceived group=0 context=from-pstn channel = 2 callerid= group= context=default ;;; line=3 WCTDM/0/2 FXSKS (SWEC: MG2) signalling=fxs_ks callerid=asreceived group=0 context=from-pstn channel = 3 callerid= group= context=default ;;; line=4 WCTDM/0/3 FXSKS (SWEC: MG2) signalling=fxs_ks callerid=asreceived group=0 context=from-pstn channel = 4 callerid= group= context=default ;;; line=5 WCTDM/0/4 FXSKS (SWEC: MG2) signalling=fxs_ks callerid=asreceived group=0 context=from-pstn channel = 5 callerid= group= context=default ;;; line=6 WCTDM/0/5 FXSKS (SWEC: MG2) signalling=fxs_ks callerid=asreceived group=0 context=from-pstn channel = 6 callerid= group= context=default ;;; line=7 WCTDM/0/6 FXSKS (SWEC: MG2) signalling=fxs_ks callerid=asreceived group=0 context=from-pstn channel = 7 callerid= group= context=default ;;; line=8 WCTDM/0/7 FXSKS (SWEC: MG2) signalling=fxs_ks callerid=asreceived group=0 context=from-pstn channel = 8 callerid= group= context=default =system.conf= # Autogenerated by /usr/sbin/dahdi_genconf on Sun Aug 22 19:34:02 2010 # If you edit this file and execute /usr/sbin/dahdi_genconf again, # your manual changes will be LOST. # Dahdi Configuration File # # This file is parsed by the Dahdi Configurator, dahdi_cfg # # Global data loadzone= us defaultzone = us # Span 1: WCTDM/0 Wildcard TDM800P Board 1 (MASTER) fxsks=1 #echocanceller=mg2,1 fxsks=2 #echocanceller=mg2,2 fxsks=3 #echocanceller=mg2,3 fxsks=4 #echocanceller=mg2,4 fxsks=5 #echocanceller=mg2,5 fxsks=6 #echocanceller=mg2,6 fxsks=7 #echocanceller=mg2,7 fxsks=8 #echocanceller=mg2,8 -- _ -- 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] Dahdi install gone wrong
They are FXO modules and yes, the lines are coming in from the telco. On 8/23/2010 12:05 PM, Doug Dawson wrote: The card you installed has FXO or FXS modules in it ? are you getting your lines directly from the telco co??? Doug D On Mon 23/08/10 8:37 AM , Cassius Smith cass...@cassius.org sent: * -Original Message- * From: Todd Reese trees...@gmail.com mailto:trees...@gmail.com * Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com mailto:asterisk-users@lists.digium.com * To: asterisk-users@lists.digium.com mailto:asterisk-users@lists.digium.com * Subject: [asterisk-users] Dahdi install gone wrong * Date: Mon, 23 Aug 2010 10:26:58 -0400 * * Hi All, * * * I've got a project installing a Digium TDM800P card with 8 FXO's in an * Asterisk box. * * * The computer is running Slackware 13.1 and I've installed the current * Dahdi and Asterisk 1.6.2.11. * * * I've installed several boxes that are pure VOIP but, I haven't installed * a Dahdi interface and I'm stumped. I've got it to the point of Dahdi * seeing the card and Asterisk recognizing dahdi but, I can't see any * channels for the calls to come in on. * * I've had to borrow files from an old config of Trixbox (the machine was * underpowered) to get to the point where I am in my setup. * * I would like to inquire some help from the group to get me up and * receiving calls on the card. * * * Regards, * * Todd Reese * * Include: * * * chan_dahdi.conf== * * * ; Configuration file * * [trunkgroups] * * [channels] * * language=en * context=from-zaptel * signalling=fxs_ks * rxwink=300 ; Atlas seems to use long (250ms) winks * ; * ; Whether or not to do distinctive ring detection on FXO lines * ; * ;usedistinctiveringdetection=yes * * usecallerid=yes * hidecallerid=no * callwaiting=yes * usecallingpres=yes * callwaitingcallerid=yes * threewaycalling=yes * transfer=yes * cancallforward=yes * callreturn=yes * echocancel=yes * echocancelwhenbridged=no * ;echotraining=800 * rxgain=0.0 * txgain=0.0 * group=0 * callgroup=1 * pickupgroup=1 * immediate=no * * ;faxdetect=both * faxdetect=incoming * ;faxdetect=outgoing * ;faxdetect=no * * ;Include setup-pstn configs * #include dahdi-channels.conf * * group=1 * * ;Include PBXconfig configs * #include chan_dahdi_additional.conf * * * * dahdi-channels.conf= * * ; Autogenerated by /usr/sbin/dahdi_genconf on Wed Aug 18 20:25:02 2010 * ; If you edit this file and execute /usr/sbin/dahdi_genconf again, * ; your manual changes will be LOST. * ; 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/chan_dahdi.conf that will include the global * settings * ; * * ; Span 1: WCTDM/0 Wildcard TDM800P Board 1 (MASTER) * ;;; line=1 WCTDM/0/0 FXSKS (SWEC: MG2) * signalling=fxs_ks * callerid=asreceived * group=0 * context=from-pstn * channel = 1 * callerid= * group= * context=default * * ;;; line=2 WCTDM/0/1 FXSKS (SWEC: MG2) * signalling=fxs_ks * callerid=asreceived * group=0 * context=from-pstn * channel = 2 * callerid= * group= * context=default * * ;;; line=3 WCTDM/0/2 FXSKS (SWEC: MG2) * signalling=fxs_ks * callerid=asreceived * group=0 * context=from-pstn * channel = 3 * callerid= * group= * context=default * * ;;; line=4 WCTDM/0/3 FXSKS (SWEC: MG2) * signalling=fxs_ks * callerid=asreceived * group=0 * context=from-pstn * channel = 4 * callerid= * group= * context=default * * ;;; line=5 WCTDM/0/4 FXSKS (SWEC: MG2) * signalling=fxs_ks * callerid=asreceived * group=0 * context=from-pstn * channel = 5 * callerid= * group= * context=default * * ;;; line=6 WCTDM/0/5 FXSKS (SWEC: MG2) * signalling=fxs_ks * callerid=asreceived * group=0 * context=from-pstn * channel = 6 * callerid= * group= * context=default * * ;;; line=7 WCTDM/0/6 FXSKS (SWEC: MG2) * signalling=fxs_ks * callerid=asreceived * group=0 * context=from-pstn * channel = 7 * callerid= * group= * context=default * * ;;; line=8 WCTDM/0/7 FXSKS (SWEC: MG2) * signalling=fxs_ks * callerid=asreceived * group=0 * context=from-pstn * channel = 8 * callerid= * group= * context=default
Re: [asterisk-users] Dahdi install gone wrong
I've made the system work by overlaying the old trixbox config in /etc/asterisk. BUT this is a disaster waiting to happen with this client. I'm having a hard time deciphering the trixbox extensions*.conf files in order to make a simple system where the client won't muck it up. On 8/23/2010 11:37 AM, Cassius Smith wrote: * -Original Message- * From: Todd Reesetrees...@gmail.com * Reply-to: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com * To: asterisk-users@lists.digium.com * Subject: [asterisk-users] Dahdi install gone wrong * Date: Mon, 23 Aug 2010 10:26:58 -0400 * * Hi All, * * * I've got a project installing a Digium TDM800P card with 8 FXO's in an * Asterisk box. * * * The computer is running Slackware 13.1 and I've installed the current * Dahdi and Asterisk 1.6.2.11. * * * I've installed several boxes that are pure VOIP but, I haven't installed * a Dahdi interface and I'm stumped. I've got it to the point of Dahdi * seeing the card and Asterisk recognizing dahdi but, I can't see any * channels for the calls to come in on. * * I've had to borrow files from an old config of Trixbox (the machine was * underpowered) to get to the point where I am in my setup. * * I would like to inquire some help from the group to get me up and * receiving calls on the card. * * * Regards, * * Todd Reese * * Include: * * * chan_dahdi.conf== * * * ; Configuration file * * [trunkgroups] * * [channels] * * language=en * context=from-zaptel * signalling=fxs_ks * rxwink=300 ; Atlas seems to use long (250ms) winks * ; * ; Whether or not to do distinctive ring detection on FXO lines * ; * ;usedistinctiveringdetection=yes * * usecallerid=yes * hidecallerid=no * callwaiting=yes * usecallingpres=yes * callwaitingcallerid=yes * threewaycalling=yes * transfer=yes * cancallforward=yes * callreturn=yes * echocancel=yes * echocancelwhenbridged=no * ;echotraining=800 * rxgain=0.0 * txgain=0.0 * group=0 * callgroup=1 * pickupgroup=1 * immediate=no * * ;faxdetect=both * faxdetect=incoming * ;faxdetect=outgoing * ;faxdetect=no * * ;Include setup-pstn configs * #include dahdi-channels.conf * * group=1 * * ;Include PBXconfig configs * #include chan_dahdi_additional.conf * * * * dahdi-channels.conf= * * ; Autogenerated by /usr/sbin/dahdi_genconf on Wed Aug 18 20:25:02 2010 * ; If you edit this file and execute /usr/sbin/dahdi_genconf again, * ; your manual changes will be LOST. * ; 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/chan_dahdi.conf that will include the global * settings * ; * * ; Span 1: WCTDM/0 Wildcard TDM800P Board 1 (MASTER) * ;;; line=1 WCTDM/0/0 FXSKS (SWEC: MG2) * signalling=fxs_ks * callerid=asreceived * group=0 * context=from-pstn * channel = 1 * callerid= * group= * context=default * * ;;; line=2 WCTDM/0/1 FXSKS (SWEC: MG2) * signalling=fxs_ks * callerid=asreceived * group=0 * context=from-pstn * channel = 2 * callerid= * group= * context=default * * ;;; line=3 WCTDM/0/2 FXSKS (SWEC: MG2) * signalling=fxs_ks * callerid=asreceived * group=0 * context=from-pstn * channel = 3 * callerid= * group= * context=default * * ;;; line=4 WCTDM/0/3 FXSKS (SWEC: MG2) * signalling=fxs_ks * callerid=asreceived * group=0 * context=from-pstn * channel = 4 * callerid= * group= * context=default * * ;;; line=5 WCTDM/0/4 FXSKS (SWEC: MG2) * signalling=fxs_ks * callerid=asreceived * group=0 * context=from-pstn * channel = 5 * callerid= * group= * context=default * * ;;; line=6
Re: [asterisk-users] Help with concurrent VoIP calls
Hi Joe, What is the app that generates your bandwidth table shown below? Joe Greco wrote: By fast I mean the best Business DSL Bellsouth has to offer: Up to 6.0 Mbps downstream - Up to 512 Kbps upstream That almost sounds like an invitation to check out what business service your cableco offers. One thing to be aware of with DSL and cable modems is that there can be various ill effects as your line gets closer to its rated capacity; do not expect that you'll get a reliable 512Kbps upstream. VoIP is sensitive to loss, latency, and jitter. You may be able, for example, to only get 384Kbps reliably out of the link (before packet loss/jitter/etc wreck its suitability for VoIP). That's a good time to look seriously at a gateway package like pfSense that can prioritize certain classes of traffic while also limiting overall bandwidth. As an example, we noticed on the local business cable offering (2Mbps up) Shaped PL min avg max stddev 2.2M3 6.4 251 557 176 2.1M1 7.8 350 584 134 2.0M3 6.4 271 535 132 1.9M1 7 254 527 131 1.8M0 6 79 339 90 1.75M 0 5.9 14 92 11 1.7M0 5.4 13 77 10 1.65M 0 4.9 11 69 7 1.6M0 5.4 13 55 9 1.5M0 5.3 11 59 7 1.4M0 5 11 57 7 1.3M0 4.9 11 54 6 1.2M0 4.9 11 52 7 1.1M0 4.8 14 53 11 The max starts trending up after 1.6M (helps to graph it) and pretty much everything goes to hell after 1.75M. ... JG ___ -- 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] Puzzling problem
Hi All, I have a problem with my Asterisk Server that the logs aren't giving me any clue to what's going on. The server is running 1.6.1.1 and connected to a Grandstream GXP2000 phone. At 3:58 minutes the call cuts off with no indication in the log. This is random and is only localized to that 1 phone. The other phone is a cordless connected through a Sipura Box with no problems. I've tried other versions of Asterisk after the problem started and it is continuing. Any help on where to look for clues is greatly appreciated. TIA, Todd Reese ___ -- 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] Puzzling problem
I did the upgrade to the phone. And the problem continued. Currently, as per the previous poster, I have reset the phone to the factory default and have started setup again. Peder wrote: Try upgrading the firmware on it. They have all sorts of goofy bugs. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Todd Reese Sent: Tuesday, June 30, 2009 4:56 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Puzzling problem Hi All, I have a problem with my Asterisk Server that the logs aren't giving me any clue to what's going on. The server is running 1.6.1.1 and connected to a Grandstream GXP2000 phone. At 3:58 minutes the call cuts off with no indication in the log. This is random and is only localized to that 1 phone. The other phone is a cordless connected through a Sipura Box with no problems. I've tried other versions of Asterisk after the problem started and it is continuing. Any help on where to look for clues is greatly appreciated. TIA, Todd Reese ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Security
Hi All, Coming in to day, the logs on the asterisk server showed several entries such as: [Apr 4 15:25:16] NOTICE[9280]: chan_sip.c:14627 handle_request_invite: Call from '' to extension '9810380487965419' rejected because extension not found. This has gotten me to thinking about security of this box. 1. Currently the box sits behind a firewall with iax and sip ports pointing to it for the ip phones that are offsite. There isn't any other access through the firewall to this box. 2. All devices have an extension assigned to them in sip.conf and extensions.conf. i.e. supra ATA, Grandstream GXP-2000 3. The box is fed via Les.net and Voicepluse. All other feeds are shutoff when not active. I'm looking for ideas to tighten up on the security so that this won't happen again. TIA, Todd Reese ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk quits responding
Hi All, Come today Dec 25, asterisk has stopped responding. I have done the usual, stop and restart Asterisk, reboot the machine, even upgraded asterisk to 1.4.23-rc3. All with the same results. Here's my symptoms, I have 2 voip phones, 2 supra ATA in the house which won't register with Asterisk. I have 2 sip providers that Asterisk won't register with. Currently from the console, Asterisk just sits there saying that it is unable to register with the remote sip providers. Give Asterisk a reload after changing one provider over to another server that I had in the config and issue a reload command, Asterisk just sits there. Type in the next command show and the display is frozen. The current config is over 2 weeks old from last update. Can someone give some ideas to look at to figure out what Asterisk is doing in this state. I have an idea that the local cable provider has monkeyed with their router config and shut down all sip users. They have done this before in the past. Thanks in advance, Todd Reese ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk quits responding
I think that it was 167. I usually keep a it very high. New update: After the console being frozen for about 15 minutes, it just responded to the last reload comand Alex Balashov wrote: What is your verbosity setting on the CLI? Try: core set verbose 10 On Dec 25, 2008, at 7:20 PM, Todd Reese trees...@gmail.com wrote: Hi All, Come today Dec 25, asterisk has stopped responding. I have done the usual, stop and restart Asterisk, reboot the machine, even upgraded asterisk to 1.4.23-rc3. All with the same results. Here's my symptoms, I have 2 voip phones, 2 supra ATA in the house which won't register with Asterisk. I have 2 sip providers that Asterisk won't register with. Currently from the console, Asterisk just sits there saying that it is unable to register with the remote sip providers. Give Asterisk a reload after changing one provider over to another server that I had in the config and issue a reload command, Asterisk just sits there. Type in the next command show and the display is frozen. The current config is over 2 weeks old from last update. Can someone give some ideas to look at to figure out what Asterisk is doing in this state. I have an idea that the local cable provider has monkeyed with their router config and shut down all sip users. They have done this before in the past. Thanks in advance, Todd Reese ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk quits responding
DNS server rebooted. Everything is back online now. What threw me was that one of the sip provider registries is by ip address and all the internal sip and ata's are by ip. That should have bypassed the DNS. Grey Man wrote: On Fri, Dec 26, 2008 at 12:36 AM, Todd Reese trees...@gmail.com wrote: I think that it was 167. I usually keep a it very high. New update: After the console being frozen for about 15 minutes, it just responded to the last reload comand Sounds like a DNS problem to me. If Asterisk can't get responses from DNS it handles it very inelegantly by just sitting there blocking. Regards, Greyman. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dropping Phone Calls
Hi All, I'm currently having trouble with dropped phone calls. The following error message is always in the log. This is a Grandstream GXP-2000 Firmware 1.1.6.16 . The Asterisk box is currently 1.4.22-rc5. The problem has been occurring on other versions also. [Sep 19 15:48:02] WARNING[13657]: chan_sip.c:1958 retrans_pkt: Maximum retries exceeded on transmission [EMAIL PROTECTED] for seqno 50706 (Critical Response) -- See doc/sip-retransmit.txt. [Sep 19 15:48:02] WARNING[13657]: chan_sip.c:1980 retrans_pkt: Hanging up call [EMAIL PROTECTED] - no reply to our critical packet (see doc/sip-retransmit.txt). Any Ideas? Regards, Todd Reese ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dropping Phone Calls
I should have added that this configuration is on a local LAN connected via a Cisco 2900 switch. On Fri, Sep 19, 2008 at 3:54 PM, Todd Reese [EMAIL PROTECTED] wrote: Hi All, I'm currently having trouble with dropped phone calls. The following error message is always in the log. This is a Grandstream GXP-2000 Firmware 1.1.6.16 . The Asterisk box is currently 1.4.22-rc5. The problem has been occurring on other versions also. [Sep 19 15:48:02] WARNING[13657]: chan_sip.c:1958 retrans_pkt: Maximum retries exceeded on transmission [EMAIL PROTECTED] for seqno 50706 (Critical Response) -- See doc/sip-retransmit.txt. [Sep 19 15:48:02] WARNING[13657]: chan_sip.c:1980 retrans_pkt: Hanging up call [EMAIL PROTECTED] - no reply to our critical packet (see doc/sip-retransmit.txt). Any Ideas? Regards, Todd Reese ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk calleri id resolution
Hi All, I'm trying to incorporate callerid_shell.agi into my Asterisk 1.4.21.2. I'm having trouble with it only returning data from the nanpa database. If I fire it up manually, I get the correct data from the sqlite3 database. What is everybody using for callerid resolution for their systems? Best Regards, Todd Reese ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk and CallerID Resolution
Hi All, I'm trying to incorporate callerid_shell.agi into my Asterisk 1.4.21.2. I'm having trouble with it only returning data from the nanpa database. If I fire it up manually, I get the correct data from the sqlite3 database. What is everybody using for callerid resolution for their systems? Best Regards, Todd Reese No virus found in this outgoing message. Checked by AVG - http://www.avg.com Version: 8.0.169 / Virus Database: 270.6.20/1666 - Release Date: 9/11/2008 7:03 AM ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 and Cepstral
Hi Russell, I, myself, have just gone through this same thing. Here is what you need to do to correct this problem. 1.Download http://www.mezzo.net/asterisk/app_swift-2.0rc1.tgz, if you haven't already. 2.Look in the source of app_swift.c for the line: const int framesize = 160*4; 3.Now change it to read: const int framesize = 20; 4.Recompile and try your new version of app_swift Regards, T. Reese - Original Message - From: Russell Handorf [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, September 17, 2007 8:56 PM Subject: [asterisk-users] Asterisk 1.4 and Cepstral Greetings, I've recently upgraded from Asterisk 1.2 to 1.4. I've been searching for a solution, but am also trying the easy way at the same time. I've now got David of Cepstral now speaking using app_swift from http://www.mezzo.net/asterisk/app_swift.html . The problem is, he sounds way worse than he did when it was asterisk version 1.2. I'm seriously considering either rolling back, or having to run a 1.2 instance connected to 1.4 via iax. What info would you all be looking for to help me trouble shoot, or are there any pointers as to what I should be looking for in a config file. Thanks. extensions.conf: exten = 403,1,Answer() exten = 403,2,Swift(Test) exten = 403,3,Swift(David^Test Test) exten = 403,4,Swift(This sounds bloody aweful) swift.conf: [general] buffer_size=65535 goto_exten=no voice=David-8kHz VOIP*CLI show application Swift VOIP*CLI -= Info about application 'Swift' =- [Synopsis] Speak text through Swift text-to-speech engine. [Description] Swift([Voice^]text) Speaks the given text through the Swift TTS engine. Returns -1 on hangup or 0 otherwise. User can exit by pressing any key. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] Cepstral's Allison is having troublespeakingclearly
Bingo! That was it. Well, it's got it to 98% there. I can play with it now and tweek it. Todd - Original Message - From: Kai-Uwe Jensen [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, September 05, 2007 12:36 AM Subject: Re: [asterisk-users] Cepstral's Allison is having troublespeakingclearly How are you playing the voice? Do you use something like app_swift or app_cepstral? Just fixed app_swift for my own installation by changing the framesize constant definition from 160*4 to 20, after googling for a similar issue. Works like a charm now. It only broke recently, i.e. not with the first 1.4.x releases, but maybe only a couple of months ago. On 9/3/07, Todd Reese [EMAIL PROTECTED] wrote: OK, I just reset the RTP packets to .020 as you have suggested. I can tell a little difference but the problem is still there. TIA, Todd -- I am Dyslexic of Borg. Fusistance is retile. Your ass will be laminated! ___ --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 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] Cepstral's Allison is having trouble speaking clearly
Hi all, I have just install and licensed Cepstral's Allison08kHz on my Asterisk 1.4.11 system. I can call the Allison's extension from my Grandstream IP Phone and she's clear as a bell, but when a call to her extension traverses through one of the Linksys/Sipura 3102 or 2002, she's got the jitters bad. The SPA-202 has only an extension phone on it and the SPA-3102 is my FXO from my Vonage Motorola box. Any clues where to start looking to clear this up? TIA, Todd Reese ___ --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] Cepstral's Allison is having troublespeaking clearly
OK, I just reset the RTP packets to .020 as you have suggested. I can tell a little difference but the problem is still there. TIA, Todd - Original Message - From: Brian West [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, September 03, 2007 6:10 PM Subject: Re: [asterisk-users] Cepstral's Allison is having troublespeaking clearly Try setting the RTP packets to 0.020 instead of 0.030 which is the default on the SPA's /b On Sep 3, 2007, at 5:00 PM, Todd Reese wrote: Hi all, I have just install and licensed Cepstral's Allison08kHz on my Asterisk 1.4.11 system. I can call the Allison's extension from my Grandstream IP Phone and she's clear as a bell, but when a call to her extension traverses through one of the Linksys/Sipura 3102 or 2002, she's got the jitters bad. The SPA-202 has only an extension phone on it and the SPA-3102 is my FXO from my Vonage Motorola box. Any clues where to start looking to clear this up? TIA, Todd Reese ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk 1.2.1 and X100 clone Zap problem
Hi all, I have just installed Asterisk 1.2.1 on my server and I'm having a problem with the X100 Zap channel. The channel works for a while when I boot up the server and then degrades to an garbled dial tone and speach. Also this problem will appear when I reload the Asterisk config files. My SIP and IAX channels are working fine. In the logs file I get Dec 26 21:05:22 WARNING[2377] chan_zap.c: Ignoring signalling Dec 26 21:05:22 WARNING[2377] chan_zap.c: Ignoring overlapdial when reloading the configs and Dec 26 21:05:41 NOTICE[2577] callerid.c: Caller*ID failed checksum Dec 26 21:05:44 NOTICE[2577] chan_zap.c: Got event 18 (Ring Begin)... when a call comes in on the Zap channel At bootup of Asterisk, these do not appear in the logs, only after a reload included zapata.conf [trunkgroups] [channels] language=en context=incoming signalling=fxs_ks channel = 1 usecallerid=yes echocancel=yes musiconhold=default immediate=no overlapdial=yes relaxdtmf=yes echotraining=yes rxgain=100% txgain=50% and zaptel.conf loadzone = us defaultzone=us fxsks=1 Any insight into resolving this would be appricated, Regards, Todd Reese ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk and cisco ubr900 configs using h.323.
I was wondering if anyone has the working configs for asterisk h323.conf andfor the cisco ubr900 voip box?TIA, Todd ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk and cisco ubr900 configs using h.323.
I was wondering if anyone has the working configs for asterisk h323.conf and for the cisco ubr900 voip box? TIA, Todd ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk, H.323 Cisco uBR900
Hi All, I have aquired a Cisco uBR900 voip router and was wondering if anyone had a working config for it and the asterisk configs. I have a reletive new verson of the CVS tree and have oh-h.323 installed. Best regards, Todd Reese ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk and h.323
Hi All, I just purchaced a Cisco uBR924 and was under the assumption that it did SIP. Being somewhat new to Asterisk, is there anyone willing to supply a working config that will get me started on configuring these items. Best Regards ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk and h.323
It's a cable modem / router. The current IOS on it is 12.1. On its back panel is a 4 port hub, 2 FXS ports, consoles and cable tv connection. - Original Message - From: Brian C. Fertig [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, July 11, 2005 6:48 PM Subject: RE: [Asterisk-Users] asterisk and h.323 To answer your question is this a router? I am not aware of this model being able to do voip. I am fluent in Cisco VOIP configs but I dont know this one. I just did some checking and this router will not do voip as far as I can tell. I believe the smallest model is a 2600 series that will do voip but your TDM voice card will cost you. they arent cheap even used. From: [EMAIL PROTECTED] on behalf of Todd Reese Sent: Mon 7/11/2005 5:42 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] asterisk and h.323 Hi All, I just purchaced a Cisco uBR924 and was under the assumption that it did SIP. Being somewhat new to Asterisk, is there anyone willing to supply a working config that will get me started on configuring these items. Best Regards ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] format g729 and Voxee.com
Hi, I have just signed up with Voxee.com and have attached my Asterisk server to dial them via IAX2. Below is the start of the log which dials the number and promply hangs up when the call is answered, with the logs saying that the channel is not compatiable. I have traced this down to the g.729 codec which I don't have installed. Any ideas on how to force that the codec not be used? BTW, I have disallow=all and allow only the codecs that I want to use in both iax.conf and sip.conf. Best Regards, Todd Reese -- Executing SetCallerID(SIP/201-fbb8, 6788896066) in new stack -- Executing Dial(SIP/201-fbb8, IAX2/134:[EMAIL PROTECTED]/17702561571) in new stack -- Called 134:[EMAIL PROTECTED]/17702561571 -- Call accepted by 66.246.246.52 (format g729) -- Format for call is g729 Jun 8 18:48:41 NOTICE[6073]: channel.c:1884 set_format: Unable to find a path from g729 to gsm Jun 8 18:48:41 NOTICE[6073]: channel.c:1884 set_format: Unable to find a path from g729 to gsm Jun 8 18:48:42 NOTICE[6073]: channel.c:1884 set_format: Unable to find a path from g729 to gsm Jun 8 18:48:42 WARNING[6405]: chan_sip.c:2170 sip_write: Asked to transmit frame type 256, while native formats is 2 (read/write = 2/2) Jun 8 18:48:42 WARNING[6405]: chan_sip.c:2170 sip_write: Asked to transmit frame type 256, while native formats is 2 (read/write = 2/2) Jun 8 18:48:42 WARNING[6405]: chan_sip.c:2170 sip_write: Asked to transmit frame type 256, while native formats is 2 (read/write = 2/2) Jun 8 18:48:51 WARNING[6405]: chan_sip.c:2170 sip_write: Asked to transmit frame type 256, while native formats is 2 (read/write = 2/2) Jun 8 18:48:51 WARNING[6405]: chan_sip.c:2170 sip_write: Asked to transmit frame type 256, while native formats is 2 (read/write = 2/2) -- IAX2/66.246.246.52:4569-7 answered SIP/201-fbb8 Jun 8 18:48:51 WARNING[6405]: channel.c:2308 ast_channel_make_compatible: No path to translate from SIP/201-fbb8(2) to IAX2/66.246.246.52:4569-7(256) Jun 8 18:48:51 WARNING[6405]: app_dial.c:1324 dial_exec_full: Had to drop call because I couldn't make SIP/201-fbb8 compatible with IAX2/66.246.246.52:4569-7 -- Hungup 'IAX2/66.246.246.52:4569-7' == Spawn extension (local-access, 17702561571, 2) exited non-zero on 'SIP/201-fbb8' ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
