[asterisk-users] HDLC Errors
0 IO-APIC-level aacraid NMI: 0 0 0 0 LOC: 742903315 742903315 742899749 742899739 ERR: 0 MIS: 0 Regards -- Daniel Johnson Systems Administrator / Systems Development Scanning Systems Australia Office: +61 7 3387 Facsimile: +61 7 3387 5588 E-mail: da...@scanningsystems.com.au Website: http://www.scanningsystems.com.au ___ -- 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] Callback / Camp / Extention Free notify?
Don Kelly wrote: You're right--he's looking for a camp-on feature, but the campon link that you found is more of a queuing feature. A properly-implemented camp-on feature has some advantages. The caller has full use of their phone while waiting for the call-back. They can make outgoing calls and receive incoming calls. They don't need to keep the handset to their ear or listen to music-on-hold in their headset. The called party won't be bothered by an incoming call unless the caller is actually prepared to talk to them (if you're waiting in-queue for 45 minutes, there is a .993 probability that the call will be answered in the three minutes while you duck out to go to the bathroom). Yes this appears to be what I am looking for. Do you know of any sites that should how to implement it correctly. As the sites that I have found are all like the voip-info wiki that Jeff sent. Or even some snipits from dialplan? Regards Daniel Johnson Systems Administrator / Systems Development Scanning Systems Australia Office: +61 7 3387 Facsimile: +61 7 3387 5588 E-mail: da...@scanningsystems.com.au Website: http://www.scanningsystems.com.au ___ -- 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] Callback / Camp / Extention Free notify?
pdha...@optusnet.com.au wrote: Daniel Johnson wrote: pdha...@optusnet.com.au wrote: Funnily enough, most people install phones with BLF lamps, on install something like hudlite/FOP/etc so you know if the person is on the phone before you call them.. PaulH Hi Paul, Yes I have seen these tools. However it is a manual process (simple, I know) and is not close to being as user friendly as the feature we are trying to achieve. Do the phones you are using support BLF? PaulH Yes, the more expensive ones do. The majority do not. Linksys phones. Its not so much knowing if the user is busy or not, its the ability to be automatically notified once the user becomes available. Daniel Johnson Systems Administrator / Systems Development Scanning Systems Australia Office: +61 7 3387 Facsimile: +61 7 3387 5588 E-mail: da...@scanningsystems.com.au Website: http://www.scanningsystems.com.au If you receive this email by mistake, please notify the sender and do not make any use of the email. We do not waive any privilege, confidentiality or copyright associated with it. ___ -- 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] Callback / Camp / Extention Free notify?
Jeff LaCoursiere wrote: I think you are looking to use a "campon" feature. Try this: http://www.voip-info.org/wiki/view/Asterisk+tips+campon j Hi Jeff, Yes I have seen this feature. Its a half implementation of what we require. The difference being that you must wait on the phone until the dialed party becomes available (be it on hold or continuous dial). >From my original email, the description of out old systems callback feature: It worked as follows. If phone A called phone B and it was BUSY, you press a button to enable a callback. User A is free to continue work or make other calls. What this meant is that when both phones became free, phone A would ring, on answer it would call phone B automatically and the call connected as per normal. Thanks for your input, Daniel Johnson Systems Administrator / Systems Development Scanning Systems Australia Office: +61 7 3387 Facsimile: +61 7 3387 5588 E-mail: da...@scanningsystems.com.au Website: http://www.scanningsystems.com.au If you receive this email by mistake, please notify the sender and do not make any use of the email. We do not waive any privilege, confidentiality or copyright associated with it. ___ -- 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] Callback / Camp / Extention Free notify?
pdha...@optusnet.com.au wrote: Funnily enough, most people install phones with BLF lamps, on install something like hudlite/FOP/etc so you know if the person is on the phone before you call them.. PaulH Hi Paul, Yes I have seen these tools. However it is a manual process (simple, I know) and is not close to being as user friendly as the feature we are trying to achieve. Daniel Johnson Systems Administrator / Systems Development Scanning Systems Australia Office: +61 7 3387 Facsimile: +61 7 3387 5588 E-mail: da...@scanningsystems.com.au Website: http://www.scanningsystems.com.au ___ -- 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] Callback / Camp / Extention Free notify?
Hi,
I am trying to implement the callback feature of our old phone system.
This feature may go by a different name in asterisk?
It worked as follows. If phone A called phone B and it was BUSY, you
press a button to enable a callback.
User A is free to continue work or make other calls.
What this meant is that when both phones became free, phone A would
ring, on answer it would call phone B automatically.
Here is how I have tried to achieve this with asterisk.
When A calls B and B is busy you get a menu, 1 to set callback and 2 to
leave voicemail.
The set of the callback is done via AGI call to PHP script to set the
details in MySQL DB.
On all call hangups I check to see if there is a pending Callback via
another AGI script.
The script sets a couple of variables which I check in my Dialplan. If
there is a callback pending for the
phone that just hung up, I need to check that the other phone involved
is FREE. This is what I can not get to work.
I have tried ChanIsAvail which does not appear to work.
I have hints setup for each SIP phone. Which would be perfect, however
it does not appear that you can check the HINT STATE in the dialplan.
I have done plenty of googling and have found this http://bugs.digium.com/view.php?id=10635
which appears that this kind of functionality was placed in 1.4.11.
(DEVICE_STATE(), EXTENSION_STATE())
I have 1.4.21.2 and do not have these features.
To continue with the rest of the feature. If the ChanIsAvail says all
is good. I then launch another AGI to write a CALLFILE and remove the
pending callback request from the DB.
This all works if A is not busy when B finishes their call etc.
I am sure that others have implemented this kind of feature. If you
could share your implementation or give me some pointers or even the
correct asterisk name so I can google and get the help I need, that
would be great.
I am considering trying out 1.6 which should have these features,
however not sure if stability is going to be a problem.
I have based my implantation based of a previous message to the list:
On Tue, Jun 10, 2008 at 5:34 PM, Phil Knighton wrote:
> Hello
>
> I'm looking for a way to do the following using my Asterisk system and Snom
> SIP phones...
>
> Scenario:
>
> Caller on Internal Phone 1 calls internal phone2. Phone 2 is busy (or more
> accurately goes straight to voicemail).
> Caller on internal phone 1 can press a button / dial a code (explained in
> next step) and hangup
> When phone 2 is free, phone 1 rings and on answer dials phone 2
>
> I was sure this was called "camping" - but all the camping stuff I can find,
> refers to the caller having to hang on the phone and wait. Am I missing
> something?
>
> Anyone have a solution?
>
Quick solution that comes into mind:
Set(exten_copy = ${EXTEN});
Dial(SIP/${EXTEN})
if ("${DIALSTATUS}"="BUSY") {
// prompt for camp
Set(DB(camp/${EXTEN}/call_to)=${CALLERID(num));
}
h => {
Set(call_to=${DB(camp/${exten_copy}/call_to)});
if ("${call_to}"!="") {
Set(DB(camp/${exten_copy}/call_to)=);
System(call_to ${exten_copy} ${call_to});
}
}
So, in case if phone2 is busy, store callerid of phone1 in database,
so when phone2 will hangup it will triger a script "call_to" which
however can originate call trough manager or call-file.
Of course you will need some additional handling in case if multiple
callers decide to camp, or diferent protocols are used, etc.
Regards,
Atis
Thanks in Advance for any help.
Regards
Daniel
Johnson
Systems Administrator / Systems Development
Scanning Systems Australia
Office: +61 7 3387
Facsimile: +61 7 3387 5588
E-mail: da...@scanningsystems.com.au
Website: http://www.scanningsystems.com.au
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[asterisk-users] Group count not being preserved when transferring a call into a conference
Hi, I am using Group and Group_Count to limit the number of calls to go out over a single peer as our channels with that peer is limited to 8. If we dial and outside number over this peer and then transfer the call into a MeetMe conference the Group gets decremented when it should not? This is most likely an error on my behalf, however I am not sure what the correct solution is. I have set the MeetMe conference up on a local extention 777. exten => 777,1,Answer exten => 777,n,MeetMe(9003|rpM) exten => 777,n,Playback(vm-goodbye) exten => 777,n,Hangup Do I need to do something in the above to preserve the Group count? Also there have been some complaints about callee's phone line being tied up and connected to the conference even after they hangup? Does meeetme not detect hangups? These calls have gone out over an IAX2 peer, is there something special I must do? Regards -- Daniel Johnson Systems Administrator / Systems Development Scanning Systems Australia Office: +61 7 3387 Facsimile: +61 7 3387 5588 E-mail: [EMAIL PROTECTED] Website: http://www.scanningsystems.com.au ___ -- 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] Parking help - causing Asterisk crash
d0", "Park Timeout -
518") in new stack
[Nov 13 15:49:09] VERBOSE[14454] logger.c: -- Executing
[EMAIL PROTECTED]:2] Set("SIP/518-b6f98fd0",
"CALLERID(name)=Park Timeout") in new stack
[Nov 13 15:49:09] VERBOSE[14454] logger.c: -- Executing
[EMAIL PROTECTED]:3] SIPAddHeader("SIP/518-b6f98fd0",
"Alert-Info: Classic-2") in new stack
[Nov 13 15:49:09] VERBOSE[14454] logger.c: -- Executing
[EMAIL PROTECTED]:4] Dial("SIP/518-b6f98fd0", "SIP/500|30") in
new stack
***CRASH***
Can someone point me in the right direction?
How should parked calls be implemented if this is wrong?
Regards
--
Daniel
Johnson
Systems Administrator / Systems Development
Scanning Systems Australia
Office: +61 7 3387
Facsimile: +61 7 3387 5588
E-mail: [EMAIL PROTECTED]
Website: http://www.scanningsystems.com.au
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Re: [asterisk-users] ZAP not answering call
Paul Hales wrote:
Just to check - have you got the right modules plugged into the right
sort of lines?
Yes - if you have a look at my zapata.conf snippet the zap chanell has
signalling=fxs_ks
Also - some analog phone interfaces are NOT standard. :(
This could be the problem. However, what I did not mention is that there
is a old Trixbox server here and it can connect up and answer the line...
I can not replicate this even if i take all the settings out of
zapata.conf and other relavant files on the tixbox.
I just wish we had a standard PSTN line here to test with, as we only
have ISDN lines.
But the line modules have to be (to work with standard phone lines, of
course)
Well these analogue lines currently just got to fax's / alarms / modems
etc. No PSTN connections
--
*Daniel Johnson*
Systems Administrator / Systems Development
Scanning Systems Australia
*Office:* +61 7 3387
*Facsimile:* +61 7 3387 5588
*E-mail:* [EMAIL PROTECTED]
<mailto:[EMAIL PROTECTED]>
*Website:* http://www.scanningsystems.com.au
PaulH
Daniel Johnson wrote:
Hi,
I am trying to interface our old PBX(Siemens) to asterisk via some
analogue ZAP lines.
The problem is that Asterisk never successfully answers the call. See
debug ouput below.
If I connect FXO -> FXS on the same card and make a call it all works
fine. So the card is not faulty.
I see there are some stange (to me) messages in the debug. I have done
search on google and tried all suggestions but they do not fix.
eg.
busydetect=no
callprogress=no
hanguponpolarityswitch=yes
Some people suggest its to do with callerID. (How can I tell if the
old PBX sends callerID?)
have tried turning callerid in asterisk on/off - changing settings. etc.
callerid=yes/no
cidstart=ring/palarity
cidsignalling=v23/bell/dtmf
Does anyone have any other ideas?
[Sep 26 12:31:54] VERBOSE[7905] logger.c: -- Starting simple
switch on 'Zap/53-1'
[Sep 26 12:31:54] NOTICE[7905] chan_zap.c: Got event 18 (Ring Begin)...
[Sep 26 12:31:55] NOTICE[7905] chan_zap.c: Got event 2 (Ring/Answered)...
[Sep 26 12:31:57] NOTICE[7905] chan_zap.c: Got event 18 (Ring Begin)...
[Sep 26 12:31:57] VERBOSE[7905] logger.c: -- Executing
[EMAIL PROTECTED]:1] Goto("Zap/53-1", "ivr-2|s|1") in new stack
[Sep 26 12:31:57] VERBOSE[7905] logger.c: -- Goto (ivr-2,s,1)
[Sep 26 12:31:57] VERBOSE[7905] logger.c: -- Executing [EMAIL PROTECTED]:1]
Set("Zap/53-1", "LOOPCOUNT=0") in new stack
[Sep 26 12:31:57] VERBOSE[7905] logger.c: -- Executing [EMAIL PROTECTED]:2]
GotoIf("Zap/53-1", "0?begin") in new stack
[Sep 26 12:31:57] VERBOSE[7905] logger.c: -- Executing [EMAIL PROTECTED]:3]
Ringing("Zap/53-1", "") in new stack
[Sep 26 12:31:57] VERBOSE[7905] logger.c: -- Executing [EMAIL PROTECTED]:4]
Answer("Zap/53-1", "") in new stack
[Sep 26 12:31:57] DEBUG[7905] chan_zap.c: Took Zap/53-1 off hook
[Sep 26 12:31:57] DEBUG[7905] chan_zap.c: No echo training requested
[Sep 26 12:31:57] VERBOSE[7905] logger.c: -- Executing [EMAIL PROTECTED]:5]
Wait("Zap/53-1", "1") in new stack
*[Sep 26 12:31:57] DEBUG[7905] chan_zap.c: Ignore switch to REVERSED
Polarity on channel 53, state 6*
*[Sep 26 12:31:57] DEBUG[7905] chan_zap.c: Ignoring Polarity switch to
IDLE on channel 53, state 6
[Sep 26 12:31:57] DEBUG[7905] chan_zap.c: Polarity Reversal event
occured - DEBUG 2: channel 53, state 6, pol= 0, aonp= 0, honp= 0,
pdelay= 600, tv= -1669362190
[Sep 26 12:31:57] WARNING[7905] chan_zap.c: Ring/Off-hook in strange
state 6 on channel 53
[Sep 26 12:31:58] DEBUG[7905] chan_zap.c: Ignore switch to REVERSED
Polarity on channel 53, state 6
[Sep 26 12:31:58] DEBUG[7905] chan_zap.c: Ignoring Polarity switch to
IDLE on channel 53, state 6
[Sep 26 12:31:58] DEBUG[7905] chan_zap.c: Polarity Reversal event
occured - DEBUG 2: channel 53, state 6, pol= 0, aonp= 0, honp= 0,
pdelay= 600, tv= -1669361310*
[Sep 26 12:31:58] VERBOSE[7905] logger.c: -- Executing [EMAIL PROTECTED]:6]
BackGround("Zap/53-1", "ssa/welcome") in new stack
[Sep 26 12:31:58] VERBOSE[7905] logger.c: -- Playing
'ssa/welcome' (language 'en')
*[Sep 26 12:31:58] WARNING[7905] chan_zap.c: Ring/Off-hook in strange
state 6 on channel 53*
[Sep 26 12:31:59] VERBOSE[7905] logger.c: -- Executing [EMAIL PROTECTED]:7]
Set("Zap/53-1", "TIMEOUT(digit)=3") in new stack
[Sep 26 12:31:59] VERBOSE[7905] logger.c: -- Digit timeout set to 3
[Sep 26 12:31:59] VERBOSE[7905] logger.c: -- Executing [EMAIL PROTECTED]:8]
Set("Zap/53-1", "TIMEOUT(response)=5") in new stack
[Sep 26 12:31:59] VERBOSE[7905] logger.c: -- Response timeout set to 5
[Sep 26 12:31:59] VERBOSE[7905] logger.c: -- Executing [EMAIL PROTECTED]:9]
WaitExten("Zap/53-1", "|") in new stack
[Sep 26 12:32:00] DEBUG[7905] chan_za
[asterisk-users] ZAP not answering call
context=from-pstn channel => 53 context=default zaptel.conf # Global data loadzone = au defaultzone = au # Span 1: WCT1/0 "Wildcard TE121 Card 0" (MASTER) B8ZS/ESF RED span=1,1,0,ccs,hdb3,crc4 bchan=1-10 unused=11-15,17,31 dchan=16 # Span 2: WCTDM/0 "Wildcard AEX2400 Board 1" fxoks=32-51 fxsks=52-55 Regards, -- *Daniel Johnson* Systems Administrator / Systems Development Scanning Systems Australia *Office:* +61 7 3387 *Facsimile:* +61 7 3387 5588 *E-mail:* [EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]> *Website:* http://www.scanningsystems.com.au ___ -- 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] PRI E1 Inbound calls hangup with busy after a few seconds
Matthew Fredrickson wrote: Daniel Johnson wrote: Hi, I have a 10 line PRI E1 ISDN service from AAPT. Connected to Asterisk 1.4 via a Digium TE121P. All oubound calls work fine. Inbound works only if I Dial a SIP phone directly or as the first step. This phone MUST NOT be busy or else the call will fail. It would appear that your zapata.conf is incorrect. One of your problems (and maybe all of them) is that it looks like you used the spanmap and trunkgroup way to setup your PRI. This is *ONLY* supposed to be used for NFAS PRIs (where you have multiple trunks per PRI). There is some tricky code in there which, when configuring using those parameters, causes it to send the channel ID in an NFAS friendly manner (which most non NFAS configured switches do not appreciate). Instead, comment out any trunkgroups or spanmaps you may have setup in zapata.conf, and do as follows: signalling=pri_ ; where pri_ is either pri_net or pri_cpe channel=1-23 ; or whatever your channels are associated with the PRI. Matthew Fredrickson Digium, Inc. Hi Matthew, Removing the trunkgroups and spanmaps has resolved this issue. Thanks *Daniel Johnson* Systems Administrator / Systems Development Scanning Systems Australia *Office:* +61 7 3387 *Facsimile:* +61 7 3387 5588 *E-mail:* [EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]> *Website:* http://www.scanningsystems.com.au ___ -- 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] PRI E1 Inbound calls hangup with busy after a few seconds
Here is the rest of the debug. > - > > PRI Debug output for successful inbound: > > > This is included in a reply as the message was to big. > < Protocol Discriminator: Q.931 (8) len=45 < Call Ref: len= 2 (reference 5493/0x1575) (Originator) < Message type: SETUP (5) < [04 03 80 90 a3] < Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) < Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16)< Sending Complete (len= 1) -- Making new call for cr 5493 -- Processing Q.931 Call Setup -- Processing IE 4 (cs0, Bearer Capability) -- Processing IE 24 (cs0, Channel Identification) -- Processing IE 108 (cs0, Calling Party Number) -- Processing IE 112 (cs0, Called Party Number) -- Processing IE 125 (cs0, High-layer Compatibility) -- Processing IE 161 (cs0, Sending Complete) q931.c:3509 q931_receive: call 5493 on channel 6 enters state 6 (Call Present) q931.c:2774 q931_call_proceeding: call 5493 on channel 6 enters state 9 (Incoming Call Proceeding) > Protocol Discriminator: Q.931 (8) len=11 > Call Ref: len= 2 (reference 5493/0x1575) (Terminator) > Message type: CALL PROCEEDING (2) > [18 04 e9 81 83 86] > Channel ID (len= 6) [ Ext: 1 IntID: Explicit PRI Spare: 0 Exclusive Dchan: 0 >ChanSel: As indicated in following octets > Ext: 1 DS1 Identifier: 1 > Ext: 1 Coding: 0 Number Specified Channel Type: 3 > Ext: 1 Channel: 6 ] -- Accepting call from '073387' to '34397333' on channel 1/6, span 1 -- Executing [EMAIL PROTECTED]:1] NoOp("Zap/6-1", "") in new stack -- Executing [EMAIL PROTECTED]:2] Dial("Zap/6-1", "SIP/511|2") in new stack -- Called 511 -- SIP/511-0821ff70 is ringing q931.c:2802 q931_alerting: call 5493 on channel 6 enters state 7 (Call Received) > Protocol Discriminator: Q.931 (8) len=9 > Call Ref: len= 2 (reference 5493/0x1575) (Terminator) > Message type: ALERTING (1) > [1e 02 81 88] > Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) > Ext: 1 Progress Description: Inband information or appropriate pattern now available. (8) ] < Protocol Discriminator: Q.931 (8) len=13 < Call Ref: len= 2 (reference 5493/0x1575) (Originator) < Message type: STATUS (125) < [08 03 80 e4 18] < Cause (len= 5) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: User (0) < Ext: 1 Cause: Invalid information element contents (100), class = Protocol Error (e.g. unknown message) (6) ] < Cause data 1: 18 (24) < [14 01 06] < Call State (len= 3) [ Ext: 0 Coding: CCITT (ITU) standard (0) Call state: Call Present (6) -- Processing IE 8 (cs0, Cause) -- Processing IE 20 (cs0, Call State) -- Nobody picked up in 2000 ms -- Executing [EMAIL PROTECTED]:3] Goto("Zap/6-1", "ConfMe|s|1") in new stack -- Goto (ConfMe,s,1) -- Executing [EMAIL PROTECTED]:1] Answer("Zap/6-1", "") in new stack q931.c:2909 q931_connect: call 5493 on channel 6 enters state 8 (Connect Request) > Protocol Discriminator: Q.931 (8) len=15 > Call Ref: len= 2 (reference 5493/0x1575) (Terminator) > Message type: CONNECT (7) > [18 04 e9 81 83 86] > Channel ID (len= 6) [ Ext: 1 IntID: Explicit PRI Spare: 0 Exclusive Dchan: 0 >ChanSel: As indicated in following octets > Ext: 1 DS1 Identifier: 1 > Ext: 1 Coding: 0 Number Specified Channel Type: 3 > Ext: 1 Channel: 6 ] > [1e 02 81 82] > Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) > Ext: 1 Progress Description: Called equipment is non-I
[asterisk-users] PRI E1 Inbound calls hangup with busy after a few seconds
ility) -- Processing IE 161 (cs0, Sending Complete) Timed out looking for connect acknowledge q931.c:2973 q931_disconnect: call 3687 on channel 7 enters state 11 (Disconnect Request) > Protocol Discriminator: Q.931 (8) len=9 > Call Ref: len= 2 (reference 3687/0xE67) (Terminator) > Message type: DISCONNECT (69) > [08 02 81 90] > Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: Private network serving the local user (1) > Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ] < Protocol Discriminator: Q.931 (8) len=5 < Call Ref: len= 2 (reference 3687/0xE67) (Originator) < Message type: RELEASE (77) q931.c:3759 q931_receive: call 3687 on channel 7 enters state 0 (Null) -- Channel 1/7, span 1 got hangup, cause -1 == Spawn extension (Inbound-Queue, s, 4) exited non-zero on 'Zap/7-1' NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Release Request > Protocol Discriminator: Q.931 (8) len=9 > Call Ref: len= 2 (reference 3687/0xE67) (Terminator) > Message type: RELEASE COMPLETE (90) > [08 02 81 90] > Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: Private network serving the local user (1) > Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ] NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null -- Hungup 'Zap/7-1' - PRI Debug output for successful inbound: This is included in a reply as the message was to big. -- Daniel Johnson Systems Administrator / Systems Development Scanning Systems Australia Office: +61 7 3387 Facsimile: +61 7 3387 5588 E-mail: [EMAIL PROTECTED] Website: http://www.scanningsystems.com.au ___ -- 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
