[asterisk-users] IP address on mysql cdr
hi, is it possible to store the IP address of the caller in the CDR? how about the end date/time? thank you. regards, ron ___ -- 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] dundi and regcontext
hi, when a user register on my asterisk i can see it adding Noop for that extension, but after awhile i won't see it anymore: what are the reasons for it being removed on the dynamic context? one thing i found when i unregister it's removed. dialplan show myregcontext [ Context 'myregcontext' created by 'SIP' ] '100500' = 1. Noop(100500) [SIP] '112802' = 1. Noop(112802) [SIP] -= 2 extensions (2 priorities) in 1 context. =- [ Context 'pfingobizsip' created by 'SIP' ] -= 0 extensions (0 priorities) in 1 context. =- my prob is when it's removed dundi cant find it anymore so a user calling from server 1 cannot call user that is in server 2. i've set re-registration to very low (1 minute) to monitor if my phone re-register and to see if it will be added again on the regcontext. but i don't even see it unregistering after 1 minute i only unregistering when i am using x-lite and closing x-lite, i dont see x-lite re-registering if i just leave the softphone open. any idea? regards, ron ___ -- 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] DUNDI Help
Hi,
I have been testing dundi setup, one thing i am having problem with is that
extensions are getting remove from the regcontext.
does it get removed when registration expires? how can i make sure it's added
back without power cycling the phone? which would be better, making expiration
higher? or lowering it so it will re-register fast? also i am using pap2 and
sipura, is there a settings to make re-register faster?
did you experience this as well before? how were you able to fix it? thank you
regards,
ron
--- On Wed, 8/27/08, Bruce Reeves [EMAIL PROTECTED] wrote:
From: Bruce Reeves [EMAIL PROTECTED]
Subject: Re: [asterisk-users] DUNDI Help
To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial
Discussion asterisk-users@lists.digium.com
Date: Wednesday, August 27, 2008, 1:06 PM
Sure, let me show you how I setup dundi on systems.
extensions.conf
exten = _1X,1,Goto(lookupdundi,${EXTEN},1)
[lookupdundi]
exten = _X,1,Goto(${ARG1},1)
switch = DUNDi/priv
exten = i,1,Playback(invalid)
You can have the i do whatever you want, and you can use the same
option in the macro you are using.
That is it, I leave out all the other context in the examples, from
time to time I add a dundi-static context and put in specific numbers
or patterns I want to accept, maybe for pstn calling or phones that
don't register, but in those cases I have multiple mappings in
dundi.conf for each context. For example:
priv = sipregistrations,0,SIP,[EMAIL PROTECTED],nopartial
priv = dundi-static,0,SIP,[EMAIL PROTECTED],nopartial
On Wed, Aug 27, 2008 at 3:56 AM, ronald ramos [EMAIL PROTECTED]
wrote:
Hi Again,
Is there a way i can detect whether a user has been added into the
regcontext?
Currently i'm seeing this and just gives a fast busy.
[Aug 27 16:44:46] WARNING[17402]: pbx.c:2483 __ast_pbx_run: Channel
'SIP/10.10.10.10-b63101d0' sent into invalid extension
'141100' in context
'lookupdundi', but no invalid handler
can i detect it somehow, so i can inform user that the extensions is not
available?
i have tried ChanIsAvail, but since i am using realtime ChanIsAvail
thinks
it registered, since it really is registered on the other server. So
it's
trying to call it, tries it for 30 secs (i set it to timeout at 30),
after 30 secs then it will go to DUNDI/priv. Is there a way that i can
detect it first so it does not try to dial it on the local before askng
dundi? thank you
regards,
Ron
--- On Tue, 8/26/08, Bruce Reeves [EMAIL PROTECTED]
wrote:
From: Bruce Reeves [EMAIL PROTECTED]
Subject: Re: [asterisk-users] DUNDI Help
To: [EMAIL PROTECTED], Asterisk Users Mailing List -
Non-Commercial
Discussion asterisk-users@lists.digium.com
Date: Tuesday, August 26, 2008, 8:16 PM
It is added when a phone registers, or re-registers. Depending on the
timing of the registrations and any restarts on the asterisk process
it may take some time for phones to re-register.
On Tue, Aug 26, 2008 at 2:10 PM, ronald ramos
[EMAIL PROTECTED]
wrote:
Hi Bruce,
my apologies, but the error was because of the key.
i just run keys init on the CLI and it works,
question
on regcontext though, i set it to sipregistrations, how often
does
an extension be added to the context sipregistrations and for how long
will
it stay there? i'm looking at dialplan show sipregistration,
sometimes
i
only see one extension there. even though i know i have 4 ip phones
registered to the asterisk.
TIA
Ron
--- On Tue, 8/26/08, Bruce Reeves [EMAIL PROTECTED]
wrote:
From: Bruce Reeves [EMAIL PROTECTED]
Subject: Re: [asterisk-users] DUNDI Help
To: [EMAIL PROTECTED], Asterisk Users Mailing List -
Non-Commercial
Discussion asterisk-users@lists.digium.com
Date: Tuesday, August 26, 2008, 6:23 PM
Ron,
What does the peers section in dundi.conf look like?
On Tue, Aug 26, 2008 at 3:00 AM, ronald
ramos
[EMAIL PROTECTED]
wrote:
Would like to try setting up dundi with 3-4 asterisk.
But for poc, i would like to try setting up dundi on between 2
asterisk.
I copied the config from DUNDI enterprise SIP with no password.
Only
thing
i
changed is the part where i used regcontext.
on both boxes dundi.conf i have
[mapping]
priv = sipregistrations,0,SIP,[EMAIL PROTECTED],nopartial
i can see both peers on each server:
CLI dundi show peers
EID HostModel AvgTime
Status
00:8e:8c:8e:cb:5310.10.10.XX (S) Symmetric Unavail OK (1
ms)
i can see my extension being added
on sipregistrations context
Added extension '136101' priority 1 to
sipregistrations
tried a dundi lookup but got no result
dundi lookup [EMAIL PROTECTED]
DUNDi lookup returned no results.
DUNDi lookup completed in 0 ms
here's what's on extensions.conf
; Private DUNDi network
[dundi-priv-canonical]
; Direct numbers
[dundi-priv-customers]
; If you are an ITSP or Reseller, list your customers here.
[dundi-priv-via-pstn]
; If you are freely delivering calls
Re: [asterisk-users] DUNDI Help
Hi Again,
Is there a way i can detect whether a user has been added into the regcontext?
Currently i'm seeing this and just gives a fast busy.
[Aug 27 16:44:46] WARNING[17402]: pbx.c:2483 __ast_pbx_run: Channel
'SIP/10..10.10.10-b63101d0' sent into invalid extension '141100' in context
'lookupdundi', but no invalid handler
can i detect it somehow, so i can inform user that the extensions is not
available?
i have tried ChanIsAvail, but since i am using realtime ChanIsAvail thinks it
registered, since it really is registered on the other server. So it's trying
to call it, tries it for 30 secs (i set it to timeout at 30), after 30 secs
then it will go to DUNDI/priv. Is there a way that i can detect it first so it
does not try to dial it on the local before askng dundi? thank you
regards,
Ron
--- On Tue, 8/26/08, Bruce Reeves [EMAIL PROTECTED] wrote:
From: Bruce Reeves [EMAIL PROTECTED]
Subject: Re: [asterisk-users] DUNDI Help
To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial
Discussion asterisk-users@lists.digium.com
Date: Tuesday, August 26, 2008, 8:16 PM
It is added when a phone registers, or re-registers. Depending on the
timing of the registrations and any restarts on the asterisk process
it may take some time for phones to re-register.
On Tue, Aug 26, 2008 at 2:10 PM, ronald ramos [EMAIL PROTECTED]
wrote:
Hi Bruce,
my apologies, but the error was because of the key.
i just run keys init on the CLI and it works,
question on regcontext though, i set it to sipregistrations, how often
does
an extension be added to the context sipregistrations and for how long
will
it stay there? i'm looking at dialplan show sipregistration, sometimes
i
only see one extension there. even though i know i have 4 ip phones
registered to the asterisk.
TIA
Ron
--- On Tue, 8/26/08, Bruce Reeves [EMAIL PROTECTED]
wrote:
From: Bruce Reeves [EMAIL PROTECTED]
Subject: Re: [asterisk-users] DUNDI Help
To: [EMAIL PROTECTED], Asterisk Users Mailing List -
Non-Commercial
Discussion asterisk-users@lists.digium.com
Date: Tuesday, August 26, 2008, 6:23 PM
Ron,
What does the peers section in dundi.conf look like?
On Tue, Aug 26, 2008 at 3:00 AM, ronald ramos
[EMAIL PROTECTED]
wrote:
Would like to try setting up dundi with 3-4 asterisk.
But for poc, i would like to try setting up dundi on between 2
asterisk.
I copied the config from DUNDI enterprise SIP with no password. Only
thing
i
changed is the part where i used regcontext.
on both boxes dundi.conf i have
[mapping]
priv = sipregistrations,0,SIP,[EMAIL PROTECTED],nopartial
i can see both peers on each server:
CLI dundi show peers
EID HostModel AvgTime Status
00:8e:8c:8e:cb:5310.10.10.XX (S) Symmetric Unavail OK (1 ms)
i can see my extension being added
on sipregistrations context
Added extension '136101' priority 1 to sipregistrations
tried a dundi lookup but got no result
dundi lookup [EMAIL PROTECTED]
DUNDi lookup returned no results.
DUNDi lookup completed in 0 ms
here's what's on extensions.conf
; Private DUNDi network
[dundi-priv-canonical]
; Direct numbers
[dundi-priv-customers]
; If you are an ITSP or Reseller, list your customers here.
[dundi-priv-via-pstn]
; If you are freely delivering calls to the PSTN, list them here
[dundi-priv-local]
include = dundi-priv-canonical
include = dundi-priv-customers
include = dundi-priv-via-pstn
[dundi-priv-switch]
; Just a wrapper for the switch
switch = DUNDi/priv
[dundi-priv-lookup]
include =
dundi-priv-local
include = dundi-priv-switch
[macro-dundi-priv]
exten = s,1,Goto(${ARG1}|1)
include = dundi-priv-lookup
[diallocal]
exten = _1X,1,Macro(dundi-priv|${EXTEN})
i also tried dialing from my xlite:
[Aug 26 15:58:07] -- Executing [EMAIL PROTECTED]:1]
Macro(SIP/138100-08269548, dundi-priv|136101)
in
new stack
[Aug 26 15:58:07] -- Executing [EMAIL PROTECTED]:1]
Goto(SIP/138100-08269548, 136101|1) in new
stack
[Aug 26 15:58:07] -- Goto (macro-dundi-priv,136101,1)
[Aug 26 15:58:07] == Auto fallthrough, channel
'SIP/138100-08269548'
status is 'UNKNOWN'
any guess what's wrong? Thanks
ron
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[asterisk-users] DUNDI Help
Would like to try setting up dundi with 3-4 asterisk.
But for poc, i would like to try setting up dundi on between 2 asterisk.
I copied the config from DUNDI enterprise SIP with no password. Only thing i
changed is the part where i used regcontext.
on both boxes dundi.conf i have
[mapping]
priv = sipregistrations,0,SIP,[EMAIL PROTECTED],nopartial
i can see both peers on each server:
CLI dundi show peers
EID Host Model AvgTime Status
00:8e:8c:8e:cb:53 10.10.10.XX (S) Symmetric Unavail OK (1 ms)
i can see my extension being added on sipregistrations context
Added extension '136101' priority 1 to sipregistrations
tried a dundi lookup but got no result
dundi lookup [EMAIL PROTECTED]
DUNDi lookup returned no results.
DUNDi lookup completed in 0 ms
here's what's on extensions.conf
; Private DUNDi network
[dundi-priv-canonical]
; Direct numbers
[dundi-priv-customers]
; If you are an ITSP or Reseller, list your customers here.
[dundi-priv-via-pstn]
; If you are freely delivering calls to the PSTN, list them here
[dundi-priv-local]
include = dundi-priv-canonical
include = dundi-priv-customers
include = dundi-priv-via-pstn
[dundi-priv-switch]
; Just a wrapper for the switch
switch = DUNDi/priv
[dundi-priv-lookup]
include = dundi-priv-local
include = dundi-priv-switch
[macro-dundi-priv]
exten = s,1,Goto(${ARG1}|1)
include = dundi-priv-lookup
[diallocal]
exten = _1X,1,Macro(dundi-priv|${EXTEN})
i also tried dialing from my xlite:
[Aug 26 15:58:07] -- Executing [EMAIL PROTECTED]:1]
Macro(SIP/138100-08269548, dundi-priv|136101) in new stack
[Aug 26 15:58:07] -- Executing [EMAIL PROTECTED]:1]
Goto(SIP/138100-08269548, 136101|1) in new stack
[Aug 26 15:58:07] -- Goto (macro-dundi-priv,136101,1)
[Aug 26 15:58:07] == Auto fallthrough, channel 'SIP/138100-08269548' status
is 'UNKNOWN'
any guess what's wrong? Thanks
ron
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Re: [asterisk-users] DUNDI Help
Hi Bruce,
my apologies, but the error was because of the key.
i just run keys init on the CLI and it works,
question on regcontext though, i set it to sipregistrations, how often does an
extension be added to the context sipregistrations and for how long will it
stay there? i'm looking at dialplan show sipregistration, sometimes i only see
one extension there. even though i know i have 4 ip phones registered to the
asterisk.
TIA
Ron
--- On Tue, 8/26/08, Bruce Reeves [EMAIL PROTECTED] wrote:
From: Bruce Reeves [EMAIL PROTECTED]
Subject: Re: [asterisk-users] DUNDI Help
To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial
Discussion asterisk-users@lists.digium.com
Date: Tuesday, August 26, 2008, 6:23 PM
Ron,
What does the peers section in dundi.conf look like?
On Tue, Aug 26, 2008 at 3:00 AM, ronald ramos [EMAIL PROTECTED]
wrote:
Would like to try setting up dundi with 3-4 asterisk.
But for poc, i would like to try setting up dundi on between 2 asterisk.
I copied the config from DUNDI enterprise SIP with no password. Only thing
i
changed is the part where i used regcontext.
on both boxes dundi.conf i have
[mapping]
priv = sipregistrations,0,SIP,[EMAIL PROTECTED],nopartial
i can see both peers on each server:
CLI dundi show peers
EID HostModel AvgTime Status
00:8e:8c:8e:cb:5310.10.10.XX (S) Symmetric Unavail OK (1 ms)
i can see my extension being added on sipregistrations context
Added extension '136101' priority 1 to sipregistrations
tried a dundi lookup but got no result
dundi lookup [EMAIL PROTECTED]
DUNDi lookup returned no results.
DUNDi lookup completed in 0 ms
here's what's on extensions.conf
; Private DUNDi network
[dundi-priv-canonical]
; Direct numbers
[dundi-priv-customers]
; If you are an ITSP or Reseller, list your customers here.
[dundi-priv-via-pstn]
; If you are freely delivering calls to the PSTN, list them here
[dundi-priv-local]
include = dundi-priv-canonical
include = dundi-priv-customers
include = dundi-priv-via-pstn
[dundi-priv-switch]
; Just a wrapper for the switch
switch = DUNDi/priv
[dundi-priv-lookup]
include = dundi-priv-local
include = dundi-priv-switch
[macro-dundi-priv]
exten = s,1,Goto(${ARG1}|1)
include = dundi-priv-lookup
[diallocal]
exten = _1X,1,Macro(dundi-priv|${EXTEN})
i also tried dialing from my xlite:
[Aug 26 15:58:07] -- Executing [EMAIL PROTECTED]:1]
Macro(SIP/138100-08269548, dundi-priv|136101) in
new stack
[Aug 26 15:58:07] -- Executing [EMAIL PROTECTED]:1]
Goto(SIP/138100-08269548, 136101|1) in new stack
[Aug 26 15:58:07] -- Goto (macro-dundi-priv,136101,1)
[Aug 26 15:58:07] == Auto fallthrough, channel
'SIP/138100-08269548'
status is 'UNKNOWN'
any guess what's wrong? Thanks
ron
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EUS Networks
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Web: www.euscorp.com
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[asterisk-users] disable auth between two asterisk
Hi, I have setup 2 asterisk talking a single mysql cluster. I'm also using realtime db. I've setup sip peering between the two asterisk servers. [asterisk-1] insecure=port,invite type=peer host=201.202.203.204 context=from-asterisk-1 [asterisk-2] insecure=port,invite type=peer host=201.202.203.205 context=from-asterisk-2 scenario: ext 100 registers on Asterisk 1 ext 200 registers on Asterisk 2. ext 100 calls ext 200. asterisk 1 receives request, asterisk 1 cannot find ext 200, forward to asterisk 2, asterisk to sends back407 proxy auth required, asterisk 1 sends proxy auth back to UA (ext 100) but i'm not sure if ext 100 is replying with the needed credentials, because asterisk 2 replies with: handle_response_invite: Failed to authenticate on INVITE to Ron sip:[EMAIL PROTECTED] i tried to disabled the password on ext 100, tried the same scenario and call went thru. so my assumption is a user registered on asterisk 1 cannot send calls to asterisk 2 coz when asterisk 2 asks for authentication, UA does not send it to asterisk 2, but i think it is sending it to asterisk 1. and vice versa if user is registered on asterisk 2, user wont be able to make calls to asterisk 1. how can i disable proxy auth on the server if the user is already registered on the other astertisk. i've set, insecure=port,invite but it still asks for proxy auth. anyone encountered this? regards, Ron ___ -- 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] how to know what codec is being used
Hi, how would i know what codec is being utilized? currently i have set allow=ilbc disallow=all. i unset all codecs on x-lite except ilbc. i tried to make a call and look at the channel i see these. does this mean it is using ulaw? how about writetranscode? does that mean there is no transcoding happening on the call? call is going thru, rtp is also going thru. what i would like to know is does it really use ilbc? i'm using 1.4.18.1. thank you core show channel SIP/19-082367b NativeFormats: 0x4 (ulaw) WriteFormat: 0x4 (ulaw) ReadFormat: 0x4 (ulaw) WriteTranscode: No ReadTranscode: No ___ -- 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] multiple asterisk approach
Hi,
I'm not sure if this is the proper way to approach it but i can't figure out
how to setup dundi.
what i did is, i try to determine which server a user is registered, by calling
an agi to query the realtime db and capture the regserver of the user.
e.g.
exten = _1xx,1,AGI(getserver.php)
exten = _1xx,2,GotoIf($[${REGSERVER} != asterisk-1]?102)
exten = _1xx,3,Dial(SIP/${EXTEN}|30|t)
exten = _1xx,102,Dial(SIP/[EMAIL PROTECTED]|30|t)
exten = _1xx,103,Hangup
then i created peering between the two. so far it is working i can call
extensions that are registered in whatever server. but what i'd like to know
is, would there be a difference on performance on calls when querying a DB to
get the regserver, or is it still adviseable to use dundi for peering.
also i setup DNS SRV for these servers, what if one server fails, should the
user close their phone to re-register to the server that is alive, or will it
automtically register to the other server if the other is unreachable? TIA
Regards
Ron
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[asterisk-users] simultaneous dial macro
Hi, Would just like to know if it's possible to be able to call a macro at the same time. i use a macro to dial local extension to another extension. exten = 100,Macro(dial-ext|SIP/100) exten = 101,Macro(dial-ext|SIP/101) but now i would like to use it on a simple ringgroup where it will ring all extensions e.g. exten = s,Dial(SIP/100SIP/101) how can i make use of my dial-ext macro instead of the simple Dial(SIP SIP SIP) thank you regards, ron ___ -- 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] simultaneous dial macro
hi,
thanks for your reply. is dialgroup already available in asterisk 1.4?
i'm currently using 1.4.21.
regards,
ron
--- On Mon, 7/28/08, Pavel Jezek [EMAIL PROTECTED] wrote:
From: Pavel Jezek [EMAIL PROTECTED]
Subject: Re: [asterisk-users] simultaneous dial macro
To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial
Discussion asterisk-users@lists.digium.com
Date: Monday, July 28, 2008, 7:52 PM
you can try to place your macro extensions into single dialgroup using
DIALGROUP() function and then reference that dialgroup in dial aplication,
eg.
Set(DIALGROUP(test,add)=Local/100)
Set(DIALGROUP(test,add)=Local/101)
Dial(${DIALGROUP(test)})
ronald ramos wrote:
Hi,
Would just like to know if it's possible to be able to call a macro at
the same time.
i use a macro to dial local extension to another extension.
exten = 100,Macro(dial-ext|SIP/100)
exten = 101,Macro(dial-ext|SIP/101)
but now i would like to use it on a simple ringgroup where it will ring
all extensions
e.g. exten = s,Dial(SIP/100SIP/101)
how can i make use of my dial-ext macro instead of the simple Dial(SIP
SIP SIP)
thank you
regards,
ron
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[asterisk-users] need help setting up dundi
Hi,
Hope anyone can help me on DUNDi. I got 2 asterisk servers. configs below.
tried this on the cli:
*CLI dundi lookup [EMAIL PROTECTED] bypass
DUNDi lookup returned no results.
DUNDi lookup completed in 0 ms
*CLI dundi lookup [EMAIL PROTECTED] bypass
DUNDi lookup returned no results.
DUNDi lookup completed in 0 ms
dundi debug shows this, i have no idea what that means though:
[Jul 24 02:42:39] Tx-Frame Retry[No] -- OSeqno: 000 ISeqno: 000 Type:
NULL (Command)
[Jul 24 02:42:39] Flags: 00 STrans: 23177 DTrans: 0 [10.10.10.1:4520]
(Final)
[Jul 24 02:42:39] Rx-Frame Retry[No] -- OSeqno: 000 ISeqno: 001 Type:
ACK (Response)
[Jul 24 02:42:39] Flags: 00 STrans: 05678 DTrans: 23177 [10.10.10.1:4520]
(Final)
any mistake on my config?
regards,
ron
asterisk#1 (IP ADDRESS:10.10.10.1)
dundi.conf
[mappings]
priv = dundi-priv-canonical,0,SIP,[EMAIL PROTECTED],nopartial
[AB:CD:EF:70:E9:DA]
model = symmetric
host = 10.10.10.2
inkey = dundi
outkey = dundi
include = priv
permit = priv
qualify = yes
order = primary
sip.conf
[4000]
type=friend
nat=yes
secret=4000
host=dynamic
[priv]
type=peer
context=dundi-priv-canonical
extensions.conf
[dundi-priv-canonical]
exten = _4XXX,1,Dial(SIP/${EXTEN})
[dundi-priv-local]
include = dundi-priv-canonical
[dundi-priv-switch]
switch = DUNDi/priv
[dundi-priv-lookup]
include = dundi-priv-local
include = dundi-priv-switch
[macro-dundi-priv]
exten = s,1,Goto(${ARG1},1)
include = dundi-priv-lookup
asterisk #2 (IP ADDRESS:10.10.10.2)
dundi.conf
[mappings]
priv = dundi-priv-canonical,0,SIP,[EMAIL PROTECTED],nopartial
[00:1E:8C:AB:CD:EF]
model = symmetric
host = 10.10.10.1
inkey = dundi
outkey = dundi
include = priv
permit = priv
qualify = yes
order = primary
sip.conf
[4001]
type=friend
nat=yes
secret=4001
host=dynamic
[priv]
type=peer
context=dundi-priv-canonical
extensions.conf
[dundi-priv-canonical]
exten = _4XXX,1,Dial(SIP/${EXTEN})
[dundi-priv-local]
include = dundi-priv-canonical
[dundi-priv-switch]
switch = DUNDi/priv
[dundi-priv-lookup]
include = dundi-priv-local
include = dundi-priv-switch
[macro-dundi-priv]
exten = s,1,Goto(${ARG1},1)
include = dundi-priv-lookup
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[asterisk-users] Play Beep if 1 minute remaining on Abosulte timeout
Hi, I have this dialpan to call international: exten =gt; _00.,1,SET(TIMEOUT(absolute)=300) exten =gt; _00.,n,Dial(SIP/[EMAIL PROTECTED]) exten =gt; _00.,n,NoCDR() exten =gt; _00.,n,Hangup Is there a way to check if there is only 1 minute remaining on the absolute timeout? also an additional question, i can make call using that dialplan, but when the remote end hangs up first, asterisk does not see the hangup so it does not disconnect the ip phone. is this a prob on my config or the gateway that i send the calls to? thank you regards ronramos ___ -- 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] trying directrtpsetup
Hi, I recently installed asterisk, i used sterisk-1.4.20.1, i i set directrtpsetup to yes, no whow would i know if the rtp/media is not passing to asterisk. any tool or can u just sniff? regards, ron ___ -- 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] install asterisk on linux that uses software raid
hi all, we recently bought a clone box, motherboard with ICH7R raid controller (which i thought was a hardware raid controller). but recently i learned that those things are called FRAID( Fake RAID) which is basically a software raid also. so i decide to just use Software RAID (using CentOS 5.1). has anyone installed asterisk on such configuration? is there any prob with regards to performance or quality of calls? thank you any info will be appreciated. regards, ron ___ -- 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] cdr question
Hi, Would just like to ask about cdr, i have an asterisk and i would like to bill only outbound calls not extension to extension, when i'm looking at the CDR, i can't figure out which fields i need to filter all outbound calls only. e.g if i dial 00. or 9XX (for local pstn calls) those are billable, 100 101 or 102 (all local extensions) not billable. *97 for voicemail not billable, but still is being logged on the cdr, can i disable logging to cdr calls like that(*98,*1,etc.)? also, the time the call ended is not logged, is there a way to log that? TIA ron - Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now.___ -- 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] ring group question
Hi All,
I'm trying to configure a ringgroup, which will ring the extension in the
group one by one. this is what i tried on my extension.conf
[macro-dial-ringgroup]
exten = s,1,Dial(SIP/${ARG1},15)
exten = s,n,NoOp( Dial Status: ${DIALSTATUS})
exten = s,n,Goto(s-${DIALSTATUS},1)
exten = s-CHANUNAVAIL,1,SetCallerId(${CALLERIDNUM})
exten = s-CHANUNAVAIL,n,Dial(SIP/${ARG1},15)
exten = s-BUSY,1,SetCallerId(${CALLERIDNUM})
exten = s-BUSY,n,Dial(SIP/${ARG1},15)
exten = s-NOANSWER,1,SetCallerId(${CALLERIDNUM})
exten = s-NOANSWER,n,Dial(SIP/${ARG1},15)
[ringgroup-1]
exten = 5000,1,Macro(dial-ringgroup,1100)
exten = 5000,n,Macro(dial-ringgroup,1101)
exten = 5000,n,Macro(dial-ringgroup,1102)
exten = 5000,n,Hangup
so when i dial 5000 it will ring 1100 no answer,or busy on 1100.
it will go to another extension which is 1101 and so on.
I have tried 5000,1,Dial(SIP/1100SIP/1100) --- this one works,
ringing at the same time, how can i do it in sequential?
hope anyone can help me. thank you
Ron
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[asterisk-users] followme scenarios
Hi All,
I'm tryng to test different scenarios for followme for different users:
[localext]
exten = 101,1,Set(FM = ALWAYS);
exten = 101,n,Macro(dial-ext|SIP/${EXTEN}|vm-1|moh-101|fm-101);
exten = 101,n,Hangup
exten = 102,1,Set(FM = NEVER);
exten = 102,n,Macro(dial-ext|SIP/${EXTEN}|vm-1|moh-102|fm-102);
exten = 102,n,Hangup
exten = 103,1,Set(FM = WHENBUSY);
exten = 103,n,Macro(dial-ext|SIP/${EXTEN}|vm-1|moh-103|fm-103);
exten = 103,n,Hangup
exten = 104,1,Set(FM = WHENUNAVAILABLE);
exten = 104,n,Macro(dial-ext|SIP/${EXTEN}|vm-1|moh-103|fm-103);
exten = 104,n,Hangup
exten = 105,1,Set(FM = CUSTOM);
exten = 105,n,Macro(dial-ext|SIP/${EXTEN}|vm-1|moh-103|fm-103);
exten = 105,n,Hangup
[macro-dial-ext]
exten = s,1,SetMusicOnHold(${ARG3})
exten = s,n,Dial(${ARG1},5,M(setmusiconhold,${ARG3}))
exten = s,n,GotoIf(FM = NEVER|?vm)
exten = s,n,GotoIf(FM = CUSTOM|?s-CUSTOM,1)
exten = s,n,GotoIf(FM = WHENUNAVAILABLE|?s-CHANUNAVAIL)
exten = s,n,GotoIf(FM = WHENBUSY|?s-BUSY)
exten = s-CHANUNAVAIL,1,Followme(${ARG4})
exten = s-BUSY,1,Followme(${ARG4})
exten = s-CUSTOM,1,GotoIftime(17:00-19:00|*|*|*?c-CUSTOM,n)
exten = s-CUSTOM,n,Followme(${ARG4})
exten = s,n,Followme(${ARG4})
exten = s,n(vm),Voicemail([EMAIL PROTECTED]|u)
exten = s,n,Playback(vm-goodbye)
exten = s,n,Hangup
but it just keeps on going to this line
exten = s,n,GotoIf(FM = NEVER|?vm)
ami using GotoIf correctly? or am i referring to the FM variable properly? and
is there easier way of doing this? TIA
regards
Ron
-
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[asterisk-users] realtime errors
Hi All, I just started playing around with asterisk realtime, added some extensions and started making test call, sometimes i can call the extension sometimes i can't. below are errors i see on the CLI, has anyone encountered this before? [settings] sippeers = mysql,sipdb,sip_customer sipusers = mysql,sipdb,sip_customer extensions = mysql,sipdb,extensions_customer voicemail = mysql,sipdb,voicemail_customer [Apr 6 01:04:53] WARNING[18959]: res_config_mysql.c:360 update_mysql: MySQL RealTime: Failed to query database. Check debug for more info. [Apr 6 01:05:04] WARNING[18959]: app_voicemail.c:2262 inboxcount: Failed to obtain database object for 'asterisk'! regards, nhadie You rock. That's why Blockbuster's offering you one month of Blockbuster Total Access, No Cost. http://tc.deals.yahoo.com/tc/blockbuster/text5.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] fax detection on sip trunk
Hi, Is it possible for me to detect fax on a sip trunk? my provider has a fax service that can send/receive fax. is it possible that i use a that trunk as a telefax? meaning i will try to detect if it's a fax, if it is i will forward it to an extension that can handle fax if not will forward it elsewhere. thank you regards ron You rock. That's why Blockbuster's offering you one month of Blockbuster Total Access, No Cost. http://tc.deals.yahoo.com/tc/blockbuster/text5.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] audio disappeared after ztdummy install
Hi All, Can't explain what happened, last night i was setting the voicemail configuration, and it worked properly: -- Executing [EMAIL PROTECTED]:3] VoiceMailMain(SIP/1000100-08219db0, @VM-1000) in new stack -- SIP/1000100-08219db0 Playing 'vm-login' (language 'en') i can hear the audio playing here. earlier i started playing with meetme, and since i don't have any zap cards, i chose to use ztdummy, -- Executing [EMAIL PROTECTED]:1] MeetMe(SIP/1000100-08206da8, 6000) in new stack == Parsing '/etc/asterisk/meetme.conf': Found -- Created MeetMe conference 1023 for conference '6000' -- SIP/1000100-08206da8 Playing 'conf-getpin' (language 'en') -- SIP/1000100-08206da8 Playing 'conf-onlyperson' (language 'en') from that message asterisk is playing conf-getpin, so i entered my conference pin number, even though i don't hear any audio, then it tried to play conf-onlyperson, still i dont hear anhything. then i tried my voicemail retrieval -- Executing [EMAIL PROTECTED]:3] VoiceMailMain(SIP/1000101-0822b6c0, @VM-1000) in new stack -- SIP/1000101-0822b6c0 Playing 'vm-login' (language 'en') same thing it's playing something but i don't hear anything. i tried playing around with my codecs, i even downloaded the alaw core and extra sound files. what do you guys think happened? it was working before i enabled ztdummy. i tested disabling the ztdummy then i can hear the audio at the voicemail but conference of course does not work now. i'm using zaptel-1.4.9.2, i tried downgrading to 1.4.8 down to 1.4.7. but still the same issue. Regards, Nhadie - Never miss a thing. Make Yahoo your homepage.___ -- 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] audio disappeared after ztdummy install
Hi, For now i just turned off acpi. and it works now. just dont know what's the connection of that though :-) i will try to do the things you guys suggested also when i get the chance, thanks for you help! regards, nhadie --- Tzafrir Cohen [EMAIL PROTECTED] wrote: On Sun, Mar 30, 2008 at 02:35:03PM -0400, Norman W. Franke wrote: All too common and largely undocumented. I had this same problem. Installing ztdummy changes Asterisk to use it for timing of playback, apparently. Removing ztdummy fixed the problem. To get it all to work, I had to upgrade to to at least kernel 2.6.23.11 (previous versions are either missing options are just broken.) Which previous versions have you tried? I'll also note that the OP needs to get Zaptel working under Xen, which is probably a different issue than your own. After doing this, I recompiled ztdummy and it worked. Note that you need to enable the various and random kernel flags to make this work, generally dealing with the high-performance timer. I enabled: HPET Timer Support Enhanced Real Time Clock Support HPET - High Precision Event Timer HPET Control RTC IRQ Allow mmap of HPET I'm not sure if you can eliminate some of those, but this works for me and is stable. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping ___ -- 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] ASTPP
Hi Sir, My problem is when I click on pricelist, i have an error there's something wrong on the pricelist database. When I looked at the database and search for a table called pricelist there's nothing there. I foolowed the querires on the the structure but also found any query that creates the pricelist table. Is the pricelist going to be created at the start or after I've setup everything? Thank You Regards, Ronald JP Carballo wrote: Under Rates click on - Pricelists then Add... ___ --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] ASTPP
Hi, Has anyone implemented astpp? I'm configuring one right now and I have a problem on the pricelist. I followed the steps here http://www.astpp.org/index.php?n=ASTPP.Installation and created tables using http://www.astpp.org/index.php?n=ASTPP.Structure, but i didn't see there a query on creating pricelist table, can anyone help me on this please? Thank You Regards, Ronald ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Prepaid Solution
Thank you all! I will check on those. Regards, Ronald JP Carballo wrote: Ronald Ramos wrote: Hi All, Any solution on how I can implement prepaid billing on asterisk? But not the calling card type, just a simple Custome rwill buy credit, consume then buy again. Also, is there a solution for that when you combine asterisk with ser? Regards, Ronald Hi Ronald, Check the prepaid applications here for ideas: http://www.voip-info.org/wiki/view/Asterisk+Prepaid+Applications ASTPP, which is based on ASTCC is highly recommended. http://www.aleph-com.net/astpp Myself, I've implemented what you aim to do using ASTCC hooked to the shopping cart Virtuemart/Joomla. Customers register through Virtuemart/Joomla, then a card is created on ASTPP. When they buy a refill card through the store, their account is credited. As for * on ser, you may want to visit : http://www.voip-info.org/wiki-SIP+Express+Router ___ --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 Prepaid Solution
Hi All, Any solution on how I can implement prepaid billing on asterisk? But not the calling card type, just a simple Custome rwill buy credit, consume then buy again. Also, is there a solution for that when you combine asterisk with ser? Regards, Ronald ___ --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] Grandstream HT486 and FAX
Hi All, I was trying to test to send a fax to an international number. Here's the setup: FAX -- HT486 -- SIP PROXY -- GATEWAY -- PSTN -- FAX Unfortunately I haven't been able to do it, I read somewhere that fax uses G711 only, is this true? because our gateway provider uses only G729. does this mean I can't send fax via that gateway, because of the codec? So can i do this then, FAX -- HT486 -- SIP PROXY -- HT486 -- FAX i'll use G711 on both HT486. Hope anyone can help me. Thanks in advanced. Regards, Ron ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
