Re: [asterisk-users] motif and other xmpp
Hi, here again, I'm going around with this problem and can not find a solution, but I put different context within xmpp.conf, asterisk believes xmpp messages between users are SIP message. any idea? 2014-11-17 16:56 GMT-06:00 ricky gutierrez : > Hi list, I have a big doubt!, I have some users with ejabberd and am > using motif to make some calls to extensions, here works fine, the > problem is when I want to send a message to another user on ejabberd > and asterisk take this message as part him, like a sip message , the > other user does not receive this message xmpp > > User A xmpp == Chat to == User B xmpp (not receive the message) > > look cli asterisk > > WARNING[20242][C-002e]: pbx.c:6646 __ast_pbx_run: Channel > 'Message/ast_msg_queue' sent to invalid extension but no invalid > handler: context,exten,priority=nica,s,1 > > any idea? > > > -- > rickygm > > http://gnuforever.homelinux.com -- rickygm http://gnuforever.homelinux.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Prohibit transfer to one extension
Hello all, first post, need help. I'm running a complex asterisk 1.8 install
with five machines. I inherited it and don't fully understand it, nor the deep
mysteries of asterisk either. I would appreciate any insight you might have. I
scoured the 'net and the Digium wiki and my Google-Fu has failed me.
I've been asked to somehow prohibit transfers to extension 3232. It has to be
fully dialable from outside, as it is now. But they want to prevent people
calling other DIDs and being transferred to 3232. They want the clerks and
other people they're calling to tell them to hang up and redial the DID for
3232. They want to be able to tell them that they CAN'T transfer them to that
extension. That extension is just an ordinary extension defined by FreePBX on
machine A. See below. Well, it's ordinary in that it was defined in the gui,
but there's no physical phone. It's just used to collect voicemail for a legal
purpose.
As I said, I inherited this system. I don't really know what to include in this
question, so I pasted anything I thought was relevant. You;ll have to tell me
if there's other config files you want to see.
I've written some VERY simple dialplan code, but need to be told where to put
it. So, here goes.
Four machines have FreePBX 2.9 front ends, the fifth is a bare asterisk install
(on machine "C").
At the same physical site as "C" is machine "A" which handles users and phones
for that site.
Other sites have machines Z, D, K to handle the phones local to them.
There's two more for the primary and failover IVR, and a subsidiary machine for
conference bridges and other minor functions.
There are (from the perspective of the "A" machine) IAX trunks to the IVR
machines, IAX trunks to Z, D, K, and "to-pstn-c" for machine C.
Machine "c" is connected to our PRIs but has no phones on it. Well, not really.
It does a few special things (in extensions.conf) like send calls to our
custom-written ACD queue program, send calls to our fax-to-print code, deal
with a few special numbers, etc. It prepends our local areacode to calls that
come in as 7-digit CLIDs. It checks if the call came in with a destination of
our main IVR, checks if it's in failover, sends it to the primary or secondary
IVR, etc.
It checks for a match against our two main DID ranges, and a few special ones:
(patterns slightly obfuscated below)
exten => _414abc32XX,1,NoOp(DID: ${EXTEN})
same => n,Goto(to-internal,${EXTEN:-4},1)
exten => _414def17XX,1,NoOP(DID: ${EXTEN})
same => n,Goto(to-internal,${EXTEN:-4},1)
exten => 414def1700,1,NoOp(Transit Plus)
same => n,Goto(1700-auto-attendant,aaentry,1)
exten => 414ghi4550,1,NoOp(Main Number)
same => n,Goto(4550-auto-attendant,aaentry,1)
exten => _X.,1,NoOp(Unknown DID: ${EXTEN})
same => n,Goto(4550-auto-attendant,aaentry,1)
exten => s,1,NoOP(No DID)
same => n,Goto(4550-auto-attendant,aaentry,1)
If there's no such DID defined, it dumps the call into our main autoattendent,
which is different from the IVR.
Right after the code above, it the [to-internal] context attempts a DUNDI
lookup on the number, and routes the call as appropriate.
AS FAR AS I CAN TELL, inbound calls are handed off to machine A, Z, D, and K,
by C, as appropriate. I think the original design was that all machines would
route calls to machine A, then A would send them to where they belonged. But
ISTR later defining meshed IAX2 trunks so that 4-digit calls could find their
way to or from A, Z, D, or K even if A was toes-up.
That's no help if machine C dies, because outbound calls to the PSTN have to go
through C.
On A, Z, D, and K, the FreePBX GUI writes the configuration files, but we do
have a BUNCH of stuff in extensions_custom.conf for special handling of various
things.
Thomas M. Peters | IT Specialist | tpet...@mcts.org
Desk: 414.343.1720 | Helpdesk: x3400 or helpd...@mcts.org
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[asterisk-users] asterisk prompt
Hello, I am trying to set up color prompt. In the documentation I have found this: %Cn[;n] Change terminal foreground (and optional background) color to specified. A full list of colors may be found in include/asterisk/term.h* But nowhere could I find what format the color code should be. I have tried all possible permutations, none of them works: ASTERISK_PROMPT="%Cn[COLOR_BLUE] %H: " asterisk -vvr ASTERISK_PROMPT="%Cn[32;128] %H: " asterisk -vvr ASTERISK_PROMPT="%Cn[32;] %H: " asterisk -vvr ASTERISK_PROMPT="%Cn[;32] %H: " asterisk -vvr ASTERISK_PROMPT="%Cn[;COLOR_CYAN] %H: " asterisk -vvr ASTERISK_PROMPT="%Cn[32|128] %H: " asterisk -vvr can somebody please tell me how to make my asterisk prompt red, for example? __ Also, I would like to piggyback a second question: I am using Asterisk 11.13 on Debian Wheezy. When I am in asterisk CLI, I can use command history and readline functions such as CTRL+r to search. But not all functions are available. For example, the alternate mappings for "page up" and "page down" to search the history do not work. They work in everything else (bash, mysql, ..) $ cat /etc/inputrc "\e[5~": history-search-forward "\e[6~": history-search-backward is there a way to make it work in asterisk ? thanks, Martin -- _ -- 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] High resident memory with 11.14.0 ?
On Tue, Nov 25, 2014 at 8:14 AM, Matthew Jordan wrote:
> On Mon, Nov 24, 2014 at 2:12 PM, James Lamanna wrote:
> > Also, how big does the cache in frame.c grow to?
> > I've recompiled with MALLOC_DEBUG on that server:
> >
> > asterisk -rx "memory show summary"
> >
> >
> > 1780466242 bytes (1780181594 cache) in2352909 allocations in file
> > frame.c
> > ...
> >
> > Seems like a ridiculous cache.
> >
>
> I'm not going to respond to your new thread, since it is the same
> discussion as this one.
>
> The frame cache is a per-thread local cache of frames that prevents
> having to re-allocate frames as they pass through Asterisk. Clearly,
> something is abusing it.
>
> I think you'll need to provide some more information on how you're
> producing this situation. Specifically:
> * Channel technologies involved, and the formats on the channels
> * Dialplan that reproduces the problem
>
> Are you using any non-core dialplan applications or channel drivers?
>
>
This PBX has about 100 registered SIP clients, along with 23 PRI channels,
2 inbound/outbound SIP trunks and around 100 IAXModems registered to it. It
primarily handles faxing.
I am not using any non-standard channel drivers. I am using the T.38
gateway funcionality.
The jist of the dialplan is this: (example of the PRI and a SIP trunk,
inbound)
[pri-in]
exten => _X.,1,Set(__FROM_DID=${EXTEN})
exten => _X.,n,Set(FAX_IDX=700)
exten => _X.,n,Set(MAX_IDX=719)
exten => _X.,n,Goto(dial-hylafax,s,1)
[sip-trunk-in]
exten => _X.,1(normal),Set(__FROM_DID=${EXTEN})
exten => _X.,n,Set(FAX_IDX=950)
exten => _X.,n,Set(MAX_IDX=959)
exten => _X.,n,Set(FAXOPT(gateway)=yes)
exten => _X.,n,Goto(dial-hylafax,s,1)
[dial-hylafax]
exten => s,1,GotoIf($["${FROM_DID:0:1}" = "1"]?prune:cont)
exten => s,n(prune),Set(__FROM_DID=${FROM_DID:1})
exten => s,n(cont),GotoIf($[${FAX_IDX} <= ${MAX_IDX}]?tryfax:nofax)
exten => s,n(tryfax),Set(STATE=${DEVICE_STATE(Custom:iaxmodem${FAX_IDX})})
exten => s,n,NoOp(${STATE})
exten => s,n,Set(DEVICE_STATE(Custom:iaxmodem${FAX_IDX})=INUSE)
exten => s,n,Dial(IAX2/iaxmodem${FAX_IDX}/${FROM_DID},60,g)
exten => s,n,Goto(s-${DIALSTATUS},1)
exten => s,n(nofax),Playtones(busy)
exten => s,n,NoOp(NO MODEMS AVAILABLE)
exten => s,n,Wait(20)
exten => s,n,Hangup()
exten => s-ANSWER,1,NoOp(IAXMODEM HANGUP)
exten => s-ANSWER,n,Set(DEVICE_STATE(Custom:iaxmodem${FAX_IDX})=NOT_INUSE)
exten => s-ANSWER,n,Hangup()
exten => _s-.,1,Set(FAX_IDX=${MATH(1+${FAX_IDX},i)})
exten => _s-.,n,Goto(s,1)
exten => h,1,Set(DEVICE_STATE(Custom:iaxmodem${FAX_IDX})=NOT_INUSE)
The current state requires me to restart Asterisk almost every day.
I'm also seeing this on a completely different machine after upgrading from
Asterisk10 to 11.
-- James
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[asterisk-users] park()-command always parks on default 701
Hello,
I have the following in my dialplan :
exten => callpark,n,Set(PARKINGDYNPOS=200-210)
exten => callpark,n,Set(PARKINGDYNCONTEXT=parked_001)
exten => callpark,n,Park(2s,parkinglot_001)
I see on the CLI :
[Nov 25 15:08:47] -- Executing [callpark@pbx-routing:10]
Set("SIP/SipT01-000b", "PARKINGDYNPOS=200-210") in new stack
[Nov 25 15:08:47] -- Executing [callpark@pbx-routing:11]
Set("SIP/SipT01-000b", "PARKINGDYNCONTEXT=parked_001") in new stack
[Nov 25 15:08:47] -- Executing [callpark@pbx-routing:12]
Park("SIP/SipT01-000b", "5s,parkinglot_001") in new stack
[Nov 25 15:08:47] == Parked SIP/SipT01-000b on 701 (lot
parkinglot_001). Will timeout back to extension [pbx-routing] s, 1 in 50
seconds
[Nov 25 15:08:47] -- Added extension '701' priority 1 to parked_77
Why does Asterisk park on 701 ? Why not on 200 ?
Kind regards,
Jonas.
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Re: [asterisk-users] High resident memory with 11.14.0 ?
On Mon, Nov 24, 2014 at 2:12 PM, James Lamanna wrote: > Also, how big does the cache in frame.c grow to? > I've recompiled with MALLOC_DEBUG on that server: > > asterisk -rx "memory show summary" > > > 1780466242 bytes (1780181594 cache) in2352909 allocations in file > frame.c > ... > > Seems like a ridiculous cache. > I'm not going to respond to your new thread, since it is the same discussion as this one. The frame cache is a per-thread local cache of frames that prevents having to re-allocate frames as they pass through Asterisk. Clearly, something is abusing it. I think you'll need to provide some more information on how you're producing this situation. Specifically: * Channel technologies involved, and the formats on the channels * Dialplan that reproduces the problem Are you using any non-core dialplan applications or channel drivers? -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com & http://asterisk.org -- _ -- 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
