Re: [asterisk-users] disconnect supervision
If you are using an analogue/sip ata, then the problem is on the ata. Run a packet capture and you'll see the invite coming from the ata without nobody using the phone... I am typing from my mobile phone... Il giorno 11/dic/2012 18:55, "Joseph" ha scritto: > On 12/11/12 11:48, Danny Nicholas wrote: > >> In /etc/asterisk/dahdi.conf, check your answeronpolarityswitch and >> hanguponpolarityswitch lines. If they aren't present, the default values >> are being used. If they are, tweak them and restart asterisk and dahdi. >> I >> do this - service asterisk stop; service dahdi restart; service asterisk >> start. >> > > I'm not using dahdi.conf I'm using extension.conf sip.conf with analog > AudioCodes gateway FXO/FXS > > -- > Joseph > > >> -Original Message- >> From: >> asterisk-users-bounces@lists.**digium.com >> [mailto:asterisk-users-**boun...@lists.digium.com] >> On Behalf Of Joseph >> Sent: Tuesday, December 11, 2012 11:44 AM >> To: Asterisk Users Mailing List - Non-Commercial Discussion >> Subject: Re: [asterisk-users] disconnect supervision >> >> On 12/11/12 11:30, Danny Nicholas wrote: >> >>> Could be, but I'd check the "easier to fix" polarity settings. >>> >> >> How do I do that? >> >> Notice, that this channel hang-up/disconnect does not happen all the time, >> only once a while could be once a day or once a week. >> >> -- >> Joseph >> > > -- > __**__**_ > -- 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<http://lists.digium.com/mailman/listinfo/asterisk-users> > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] disconnect supervision
You need to look at the device which the analog lines plug into. There is nothing to change in Asterisk for this issue. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joseph Sent: Tuesday, December 11, 2012 12:56 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] disconnect supervision On 12/11/12 11:48, Danny Nicholas wrote: >In /etc/asterisk/dahdi.conf, check your answeronpolarityswitch and >hanguponpolarityswitch lines. If they aren't present, the default >values are being used. If they are, tweak them and restart asterisk >and dahdi. I do this - service asterisk stop; service dahdi restart; >service asterisk start. I'm not using dahdi.conf I'm using extension.conf sip.conf with analog AudioCodes gateway FXO/FXS -- Joseph > >-Original Message- >From: asterisk-users-boun...@lists.digium.com >[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joseph >Sent: Tuesday, December 11, 2012 11:44 AM >To: Asterisk Users Mailing List - Non-Commercial Discussion >Subject: Re: [asterisk-users] disconnect supervision > >On 12/11/12 11:30, Danny Nicholas wrote: >>Could be, but I'd check the "easier to fix" polarity settings. > >How do I do that? > >Notice, that this channel hang-up/disconnect does not happen all the >time, only once a while could be once a day or once a week. > >-- >Joseph -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] disconnect supervision
On 12/11/12 11:48, Danny Nicholas wrote: In /etc/asterisk/dahdi.conf, check your answeronpolarityswitch and hanguponpolarityswitch lines. If they aren't present, the default values are being used. If they are, tweak them and restart asterisk and dahdi. I do this - service asterisk stop; service dahdi restart; service asterisk start. I'm not using dahdi.conf I'm using extension.conf sip.conf with analog AudioCodes gateway FXO/FXS -- Joseph -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joseph Sent: Tuesday, December 11, 2012 11:44 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] disconnect supervision On 12/11/12 11:30, Danny Nicholas wrote: Could be, but I'd check the "easier to fix" polarity settings. How do I do that? Notice, that this channel hang-up/disconnect does not happen all the time, only once a while could be once a day or once a week. -- Joseph -- _ -- 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] disconnect supervision
In /etc/asterisk/dahdi.conf, check your answeronpolarityswitch and hanguponpolarityswitch lines. If they aren't present, the default values are being used. If they are, tweak them and restart asterisk and dahdi. I do this - service asterisk stop; service dahdi restart; service asterisk start. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joseph Sent: Tuesday, December 11, 2012 11:44 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] disconnect supervision On 12/11/12 11:30, Danny Nicholas wrote: >Could be, but I'd check the "easier to fix" polarity settings. How do I do that? Notice, that this channel hang-up/disconnect does not happen all the time, only once a while could be once a day or once a week. -- Joseph -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] disconnect supervision
On 12/11/12 11:30, Danny Nicholas wrote: Could be, but I'd check the "easier to fix" polarity settings. How do I do that? Notice, that this channel hang-up/disconnect does not happen all the time, only once a while could be once a day or once a week. -- Joseph -- _ -- 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] disconnect supervision
Could be, but I'd check the "easier to fix" polarity settings. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joseph Sent: Tuesday, December 11, 2012 11:27 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] disconnect supervision Most of the time my phone line are working OK but at time to time when I run: asterisk -rx "core show channels" it show: Channel Location State Application(Data) SIP/pstn--00 (None) Up AppDial((Outgoing Line)) SIP/pstn-9998-00 7807586576@internal: Up Dial(SIP/97807807586576@pstn-4 2 active channels 1 active call even though nobody is using any line. I'm using Audiocodes gateway. Does it have anything to do with "disconnect supervision" on analog line in Audiocodes gateway? -- Joseph -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] disconnect supervision
Most of the time my phone line are working OK but at time to time when I run: asterisk -rx "core show channels" it show: Channel Location State Application(Data) SIP/pstn--00 (None) Up AppDial((Outgoing Line)) SIP/pstn-9998-00 7807586576@internal: Up Dial(SIP/97807807586576@pstn-4 2 active channels 1 active call even though nobody is using any line. I'm using Audiocodes gateway. Does it have anything to do with "disconnect supervision" on analog line in Audiocodes gateway? -- Joseph -- _ -- 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] Disconnect supervision tone detection working for india
Hi ,
Thanks danny nicholas. Finally we get the things done with following.
If i specify busypatten=500,500 then asterisk does not recognize hang up
signal. After removing it only all are working fine.
I choosed 2nd option as per your suggestions.
working chan-dahdi.conf:
signalling = fxs_ks
busycount = 3
busydetect = yes
callprogress = yes
progzone=in
usecallerid=yes
cidstart=ring
callerid=asreceived
group=0
context=from-pstn
channel => 1
Caller id Detection :
I have PSTN line with caller id display. After the 2nd ring, caller id
display phone shows caller id. Can you guide me on right to get it in
asterisk.
I am getting following error when line is ringing ,
*CLI> -- Starting simple switch on 'DAHDI/1-1'
[Jul 31 15:26:24] ERROR[5216]: callerid.c:564 callerid_feed: No start bit
found in fsk data.
[Jul 31 15:26:24] WARNING[5216]: chan_dahdi.c:7016 ss_thread: Failed to
decode CallerID on channel 'DAHDI/1-1'
[Jul 31 15:26:24] WARNING[5216]: chan_dahdi.c:7121 ss_thread: CallerID
returned with error on channel 'DAHDI/1-1'
-- Executing [...@from-pstn:1] AGI("DAHDI/1-1", "agi://localhost") in new
stack
-- AGI Script Executing Application: (BACKGROUND) Options:
(/home/guest/adhearsion/songs_play/voices/welcome)
-- Playing
'/home/guest/adhearsion/songs_play/voices/welcome' (language 'en')
hope guidance from you,
Ashik
On Sat, Jul 31, 2010 at 1:50 PM, asteriskguru asteriskguru <
beaasteriskg...@gmail.com> wrote:
> hi,
>
> Although I changed those parameters. Asterisk does not detect "hangup
> signal"
>
>
> signalling = fxs_ks
> busydetect = yes
> busycount = 3
> busypattern = 500,500
> answeronpolarityswitch = no
> hanguponpolarityswitch = yes
>
> callprogress=no
> progzone=in
> usecallerid=yes
> cidstart=ring
> callerid=asreceived
> group=0
> context=from-pstn
> channel => 1
>
> by hearing attached tone anything can be done ?
>
> hope guidance from you,
> ashik
>
>
> On Fri, Jul 30, 2010 at 7:20 PM, Danny Nicholas wrote:
>
>> Your best bets are going to be
>>
>> #1 hanguponpolarityswitch=yes
>>
>> Or
>>
>> #2 callprogress=yes
>>
>>
>>
>> I’d hang my hat on #1 personally
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>> http://www.asterisk.org/hello
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Disconnect supervision tone detection
Hi ,
Thanks danny nicholas. Finally we get the things done with following.
If i specify busypatten=500,500 then asterisk does not recognize hang up
signal. After removing it only all are working fine.
I choosed 2nd option as per your suggestions.
working chan-dahdi.conf:
signalling = fxs_ks
busycount = 3
busydetect = yes
callprogress = yes
progzone=in
usecallerid=yes
cidstart=ring
callerid=asreceived
group=0
context=from-pstn
channel => 1
Caller id Detection :
I have PSTN line with caller id display. After the 2nd ring, caller id
display phone shows caller id. Can you guide me on right to get it in
asterisk.
I am getting following error when line is ringing ,
*CLI> -- Starting simple switch on 'DAHDI/1-1'
[Jul 31 15:26:24] ERROR[5216]: callerid.c:564 callerid_feed: No start bit
found in fsk data.
[Jul 31 15:26:24] WARNING[5216]: chan_dahdi.c:7016 ss_thread: Failed to
decode CallerID on channel 'DAHDI/1-1'
[Jul 31 15:26:24] WARNING[5216]: chan_dahdi.c:7121 ss_thread: CallerID
returned with error on channel 'DAHDI/1-1'
-- Executing [...@from-pstn:1] AGI("DAHDI/1-1", "agi://localhost") in new
stack
-- AGI Script Executing Application: (BACKGROUND) Options:
(/home/guest/adhearsion/songs_
play/voices/welcome)
-- Playing
'/home/guest/adhearsion/songs_play/voices/welcome' (language 'en')
hope guidance from you,
Ashik
On Sat, Jul 31, 2010 at 1:50 PM, asteriskguru asteriskguru <
beaasteriskg...@gmail.com> wrote:
> hi,
>
> Although I changed those parameters. Asterisk does not detect "hangup
> signal"
>
>
> signalling = fxs_ks
> busydetect = yes
> busycount = 3
> busypattern = 500,500
> answeronpolarityswitch = no
> hanguponpolarityswitch = yes
>
> callprogress=no
> progzone=in
> usecallerid=yes
> cidstart=ring
> callerid=asreceived
> group=0
> context=from-pstn
> channel => 1
>
> by hearing attached tone anything can be done ?
>
> hope guidance from you,
> ashik
>
>
> On Fri, Jul 30, 2010 at 7:20 PM, Danny Nicholas wrote:
>
>> Your best bets are going to be
>>
>> #1 hanguponpolarityswitch=yes
>>
>> Or
>>
>> #2 callprogress=yes
>>
>>
>>
>> I’d hang my hat on #1 personally
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>> http://www.asterisk.org/hello
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Disconnect supervision tone detection
hi, Although I changed those parameters. Asterisk does not detect "hangup signal" signalling = fxs_ks busydetect = yes busycount = 3 busypattern = 500,500 answeronpolarityswitch = no hanguponpolarityswitch = yes callprogress=no progzone=in usecallerid=yes cidstart=ring callerid=asreceived group=0 context=from-pstn channel => 1 by hearing attached tone anything can be done ? hope guidance from you, ashik On Fri, Jul 30, 2010 at 7:20 PM, Danny Nicholas wrote: > Your best bets are going to be > > #1 hanguponpolarityswitch=yes > > Or > > #2 callprogress=yes > > > > I’d hang my hat on #1 personally > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Disconnect supervision tone detection
Your best bets are going to be #1 hanguponpolarityswitch=yes Or #2 callprogress=yes I'd hang my hat on #1 personally -- _ -- 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] Disconnect supervision tone detection
Hi, I changed all settings in users.conf to chan-dahdi.conf here is configuration settings. chan_dahdi.conf: -- signalling=fxs_ks busydetect=yes busycount=3 busypattern=300,300 answeronpolarityswitch = no hanguponpolarityswitch = no callprogress = no progzone = in usecallerid = yes cidstart = ring callerid=asreceived group=0 context=from-pstn channel => 1 ;;; line="2 WCTDM/4/1 FXOKS" signalling=fxo_ks callerid="Channel 2" <4002> mailbox=4002 group=5 context=from-internal channel => 2 ;;; line="3 WCTDM/4/2 FXOKS" signalling=fxo_ks callerid="Channel 3" <4003> mailbox=4003 group=5 context=from-internal channel => 3 ;;; line="4 WCTDM/4/3 FXOKS" signalling=fxo_ks callerid="Channel 4" <4004> mailbox=4004 group=5 context=from-internal channel => 4 *CLI> dahdi show channel 1 Channel: 1 File Descriptor: 16 Span: 1 Extension: Dialing: no Context: from-pstn Caller ID: Calling TON: 0 Caller ID name: Destroy: 0 InAlarm: 0 Signalling Type: FXS Kewlstart Radio: 0 Owner: Real: Callwait: Threeway: Confno: -1 Propagated Conference: -1 Real in conference: 0 DSP: no Relax DTMF: no Dialing/CallwaitCAS: 0/0 Default law: ulaw Fax Handled: no Pulse phone: no Echo Cancellation: 128 taps, currently OFF Actual Confinfo: Num/0, Mode/0x Actual Confmute: No Hookstate (FXS only): Offhook === Hope guidance from you, Ashik On Fri, Jul 30, 2010 at 12:22 AM, Danny Nicholas wrote: > We will need to see your dahdi.conf and chan_dahdi.conf files as well. > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Disconnect supervision tone detection
We will need to see your dahdi.conf and chan_dahdi.conf files as well. -- _ -- 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] Disconnect supervision tone detection
Hi, I am using TDM400 card with 3 fxs and 1 fxo. I am struggling to detect hangup tone or disconnect supervision tone from my CO. I attached the recorded wav file which contains my telco's disconnect supervision. I am using , asterisk-1.4.33.1 dahdi-linux-complete-2.3.0.1+ 2.3.0 OS => Debian-lenny 5 users.conf - [trunk_1] trunkname = pstn ; GUI metadata busydetect = yes busycount = 3 busypattern = 480,620 ringtimeout = 8000 answeronpolarityswitch = no hanguponpolarityswitch = no callprogress = no progzone = in usecallerid = yes cidstart = ring pulsedial = no cidsignalling = v23 flash = 750 rxflash = 1250 mailbox = callerid = asreceived dahdichan = 1 context = DID_trunk_1 group = 1 hasexten = no hasiax = no hassip = no registeriax = no registersip = no trunkstyle = analog disallow = all allow = all gui_volume = 2 ; GUI metadata signalling = fxs_ks gui_fxooffset = 0 ; GUI metadata rxgain = 0 txgain = 0.0 channel = 1 /dahdi/system.conf # Span 1: WCTDM/4 "Wildcard TDM400P REV I Board 5" (MASTER) fxsks=1 fxoks=2 fxoks=3 fxoks=4 # Global data loadzone= in defaultzone = in indications.conf: [general] country=in ; default location [in] description = India ringcadence = 400,200,400,2000 dial = 400*25 busy = 400/750,0/750 ring = 400*25/400,0/200,400*25/400,0/2000 congestion = 400/250,0/250 callwaiting = 400/200,0/100,400/200,0/7500 dialrecall = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440 record = 1400/500,0/15000 info = !950/330,!1400/330,!1800/330,0/1000 stutter = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440 = Can anybody guide me on right ? . although I spent 1 day I didn't get any answers for this. I am keen on to write blog for this if it is works well. It will helpfull for india asterisk users. hope guidance from you , as...@india -- _ -- 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] Disconnect Supervision UK / BT solution?
Chris Earle (CBL) wrote: Sorry -- you're right, I didn't express the scenario properly .. The disconnect supervision problem is when I 'forward'/divert an incoming POTS call out another FXO channel to a mobile phone or POTS line. (POTS -> Sangoma|Asterisk -> POTS/mobile) When the incoming POTS hangs up and/or the mobile the person was connected to .. Asterisk/Sangoma doesn't hang the Zap channels up. Just to clarifydoes it all work ok if you are using SIP or IAX for the forwarded channels? Eg local SIP phones? I only have incoming zap lines in my config and with the exception of hangup on ringing I have found hangup detection to work fine. I have a fax machine forwarding in my config as well and again no problems yet with hangup on that. Does it fail to work *every* time, or just intermittently? Does CallerId work ok in your setup? (can be a clue to help diagnose your setup) Ed W ___ --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] Disconnect Supervision UK / BT solution?
Sorry -- you're right, I didn't express the scenario properly .. The disconnect supervision problem is when I 'forward'/divert an incoming POTS call out another FXO channel to a mobile phone or POTS line. (POTS -> Sangoma|Asterisk -> POTS/mobile) When the incoming POTS hangs up and/or the mobile the person was connected to .. Asterisk/Sangoma doesn't hang the Zap channels up. I have tried busydetect and busycounts and a number of settings are enabled for UK CallerID support (polarity switch stuff) ... but I had some sketchy side effects with busydetect etc and am wary of premature hangups Thanks for your query -- Chris - Original Message - From: "Ed W" <[EMAIL PROTECTED]> To: "Chris Earle (CBL)" <[EMAIL PROTECTED]>; "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Friday, January 19, 2007 1:26 PM Subject: Re: [asterisk-users] Disconnect Supervision UK / BT solution? > > > Does anyone have any thoughts/confirmation about this finally being a viable > > solution? This disconnect supervision problem has plagued TDM and Sangoma > > cards for a long time! > > > > Just to be clear, what is the exact "disconnect problem" that you see? > > I have three TDM cards in two different systems, one using PBX lines and > one on a private BT line. Both of them have trouble detecting a caller > who is ringing, but then hangs up before being answered by the asterisk > system > > However, *all* of them are absolutely fine at spotting a normal hangup > once the call is connected and I see no random disconnects during calls > either. > > Can you confirm that this is what you mean, or whether it's something else? > > Ed W > > > -- > This message has been scanned for viruses and > dangerous content and is believed to be clean. -- This message has been scanned for viruses and dangerous content and is believed to be clean. ___ --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] Disconnect Supervision UK / BT solution?
thanks for your helpful investigation! I await news :-) -- Chris - Original Message - From: "Matt Brown" <[EMAIL PROTECTED]> To: "Chris Earle (CBL)" <[EMAIL PROTECTED]>; "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Saturday, January 20, 2007 7:55 AM Subject: Re: [asterisk-users] Disconnect Supervision UK / BT solution? > Well, > > I have just phoned BT today who said they can increase the CPC value > on the line - however it needs to be done at the exchange - and has > been booked for Tues. > > I suppose I will know wether this worked on Tues :-) - I shall post > my findings. > > Regards > > -- > Matt Brown > > > > On 19 Jan 2007, at 16:46, Chris Earle ((CBL)) wrote: > > > Hi all > > > > I'm using sangoma a200 cards in the UK and have the ongoing, often > > noted > > problem of disconnect supervision with BT POTS lines. > > > > Just noticed this post on > > http://www.voip-info.org/wiki/view/UK+Asterisk+Details > > stating that potentially someone's got a solution : > > > > "TDM400P & Not Detecting Hangups: > > > > Got a TDM400P installed and having problems with Asterisk not > > detecting > > hangups? Using BT? If so, contact BT and ask what the "Disconnect > > Clear > > Time" setting is for your phone line. Odds are it's probably 100. > > Increasing > > it to 800 fixed the issue for me. > > > > "Disconnect Clear Time" is BT's name for CPC. " > > > > > > Does anyone have any thoughts/confirmation about this finally being > > a viable > > solution? This disconnect supervision problem has plagued TDM and > > Sangoma > > cards for a long time! > > > > Comments appreciated before I get on the phone with BT > > > > > > -- > > Chris Earle > > System Solutions Specialist > > > > > > -- > > This message has been scanned for viruses and > > dangerous content and is believed to be clean. > > > > ___ > > --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 > > > -- > This message has been scanned for viruses and > dangerous content and is believed to be clean. -- This message has been scanned for viruses and dangerous content and is believed to be clean. ___ --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] Disconnect Supervision UK / BT solution?
Matt Brown wrote: Well, I have just phoned BT today who said they can increase the CPC value on the line - however it needs to be done at the exchange - and has been booked for Tues. I suppose I will know wether this worked on Tues :-) - I shall post my findings. I would be keen to hear your findings - however, I'm still not clear exactly what the "problem" is in your case. There are numerous kinds of disconnect problems - which one are you having (so we know which one the CPC fixes...) Cheers Ed W ___ --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] Disconnect Supervision UK / BT solution?
Hello, What's your zapata.conf and zaptel.conf? On 1/20/07, Matt Brown <[EMAIL PROTECTED]> wrote: Well, I have just phoned BT today who said they can increase the CPC value on the line - however it needs to be done at the exchange - and has been booked for Tues. I suppose I will know wether this worked on Tues :-) - I shall post my findings. Regards -- Matt Brown On 19 Jan 2007, at 16:46, Chris Earle ((CBL)) wrote: > Hi all > > I'm using sangoma a200 cards in the UK and have the ongoing, often > noted > problem of disconnect supervision with BT POTS lines. > > Just noticed this post on > http://www.voip-info.org/wiki/view/UK+Asterisk+Details > stating that potentially someone's got a solution : > > "TDM400P & Not Detecting Hangups: > > Got a TDM400P installed and having problems with Asterisk not > detecting > hangups? Using BT? If so, contact BT and ask what the "Disconnect > Clear > Time" setting is for your phone line. Odds are it's probably 100. > Increasing > it to 800 fixed the issue for me. > > "Disconnect Clear Time" is BT's name for CPC. " > > > Does anyone have any thoughts/confirmation about this finally being > a viable > solution? This disconnect supervision problem has plagued TDM and > Sangoma > cards for a long time! > > Comments appreciated before I get on the phone with BT > > > -- > Chris Earle > System Solutions Specialist > > > -- > This message has been scanned for viruses and > dangerous content and is believed to be clean. > > ___ > --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 ___ --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 ___ --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] Disconnect Supervision UK / BT solution?
Well, I have just phoned BT today who said they can increase the CPC value on the line - however it needs to be done at the exchange - and has been booked for Tues. I suppose I will know wether this worked on Tues :-) - I shall post my findings. Regards -- Matt Brown On 19 Jan 2007, at 16:46, Chris Earle ((CBL)) wrote: Hi all I'm using sangoma a200 cards in the UK and have the ongoing, often noted problem of disconnect supervision with BT POTS lines. Just noticed this post on http://www.voip-info.org/wiki/view/UK+Asterisk+Details stating that potentially someone's got a solution : "TDM400P & Not Detecting Hangups: Got a TDM400P installed and having problems with Asterisk not detecting hangups? Using BT? If so, contact BT and ask what the "Disconnect Clear Time" setting is for your phone line. Odds are it's probably 100. Increasing it to 800 fixed the issue for me. "Disconnect Clear Time" is BT's name for CPC. " Does anyone have any thoughts/confirmation about this finally being a viable solution? This disconnect supervision problem has plagued TDM and Sangoma cards for a long time! Comments appreciated before I get on the phone with BT -- Chris Earle System Solutions Specialist -- This message has been scanned for viruses and dangerous content and is believed to be clean. ___ --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 ___ --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] Disconnect Supervision UK / BT solution?
Does anyone have any thoughts/confirmation about this finally being a viable solution? This disconnect supervision problem has plagued TDM and Sangoma cards for a long time! Just to be clear, what is the exact "disconnect problem" that you see? I have three TDM cards in two different systems, one using PBX lines and one on a private BT line. Both of them have trouble detecting a caller who is ringing, but then hangs up before being answered by the asterisk system However, *all* of them are absolutely fine at spotting a normal hangup once the call is connected and I see no random disconnects during calls either. Can you confirm that this is what you mean, or whether it's something else? Ed W ___ --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] Disconnect Supervision UK / BT solution?
Hi all I'm using sangoma a200 cards in the UK and have the ongoing, often noted problem of disconnect supervision with BT POTS lines. Just noticed this post on http://www.voip-info.org/wiki/view/UK+Asterisk+Details stating that potentially someone's got a solution : "TDM400P & Not Detecting Hangups: Got a TDM400P installed and having problems with Asterisk not detecting hangups? Using BT? If so, contact BT and ask what the "Disconnect Clear Time" setting is for your phone line. Odds are it's probably 100. Increasing it to 800 fixed the issue for me. "Disconnect Clear Time" is BT's name for CPC. " Does anyone have any thoughts/confirmation about this finally being a viable solution? This disconnect supervision problem has plagued TDM and Sangoma cards for a long time! Comments appreciated before I get on the phone with BT -- Chris Earle System Solutions Specialist -- This message has been scanned for viruses and dangerous content and is believed to be clean. ___ --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
Fwd: [asterisk-users] Disconnect supervision in India?
On 1/1/07, ram <[EMAIL PROTECTED]> wrote: On 12/30/06, Rajkumar S <[EMAIL PROTECTED]> wrote: > On 12/29/06, Chris Earle <[EMAIL PROTECTED]> wrote: > > anyone know the status of disconnect supervision on POTS lines in India? > > Set up an asterisk box, TDM cards, in Mumbai, and doesn't seem to have > > disconnect supervision.. > > It does not work afaik, you may not get caller id also. I tested upto > 1.4b3 and no luck. its all depends on the provider where you take from. Does any provider's land line works well with TDM Cards? raj ___ --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] Disconnect supervision in India?
On 12/30/06, Rajkumar S <[EMAIL PROTECTED]> wrote: On 12/29/06, Chris Earle <[EMAIL PROTECTED]> wrote: > anyone know the status of disconnect supervision on POTS lines in India? > Set up an asterisk box, TDM cards, in Mumbai, and doesn't seem to have > disconnect supervision.. It does not work afaik, you may not get caller id also. I tested upto 1.4b3 and no luck. raj its all depends on the provider where you take from. ram ___ --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] Disconnect supervision in India?
On 12/29/06, Chris Earle <[EMAIL PROTECTED]> wrote: anyone know the status of disconnect supervision on POTS lines in India? Set up an asterisk box, TDM cards, in Mumbai, and doesn't seem to have disconnect supervision.. It does not work afaik, you may not get caller id also. I tested upto 1.4b3 and no luck. raj ___ --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] Disconnect supervision in India?
Hey all, anyone know the status of disconnect supervision on POTS lines in India? Set up an asterisk box, TDM cards, in Mumbai, and doesn't seem to have disconnect supervision.. Thanks -- Chris Earle System Solutions Specialist ___ --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] Disconnect supervision question
> Make sure your PBX/KSU is giving a disconnect indication that > asterisk understands (battery drop or battery reversal). If > not, there's not a damn thing you can do. That's what I'm starting to suspect. If I dial the Asterisk machine as a local extension my PBX will send a busy signal when the line is dropped and Asterisk seems to detect it with busy + progress enabled, but when an outside caller is forwarded to the extension it doesn't get the busy-on-drop. There's a few settings I can try in the PBX (an Iwatsu ZTD). ___ 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] Disconnect supervision question
On Thursday 18 August 2005 12:13, James Fogg wrote: > The interface is a non-Digium single port FXO card (modem) with R13 & > R19 removed to mimic the Digium card. The software is the current > release of [EMAIL PROTECTED] from ISO format (self booting CD that does an > OS install and Asterisk compile automagically). Make sure your PBX/KSU is giving a disconnect indication that asterisk understands (battery drop or battery reversal). If not, there's not a damn thing you can do. -A. ___ 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] Disconnect supervision question
Answering machines are a lot better at detecting hangups than Asterisk is. Asterisk expects either a removal of voltage from the line or a reverse of polarity of the line. BJ Weschke wrote: Your line must provide disconnect supervision, which it sound like you say that it does, and you must configure that Zap trunk in asterisk for "kewl start" signaling (fxoks) which makes use of the disconnect supervision. On 8/18/05, James Fogg <[EMAIL PROTECTED]> wrote: I'm trying to use Asterisk as a voice mail system and all is going well except for one issue. When the outside caller drops the connection Asterisk doesn't sense it. It keeps the line open for about 2 minutes before dropping it. The FXO is attached to a key system (hybrid PBX). There was a home-type digital answering machine previously on the same extension and it was able to sense disconnect without any problems. The interface is a non-Digium single port FXO card (modem) with R13 & R19 removed to mimic the Digium card. The software is the current release of [EMAIL PROTECTED] from ISO format (self booting CD that does an OS install and Asterisk compile automagically). My question is if Asterisk has settings to control disconnect sense, or do I have to look at the key system (hybrid PBX) that it's attached to. -- Eric Wieling * BTEL Consulting * 504-210-3699 x2120 r: Generate a ringing tone for the calling party, passing no audio from the called channel(s) until one answers. Use with care and don't insert this by default into all your dial statements as you are killing call progress information for the user. Really, you almost certainly do not want to use this. Asterisk will generate ring tones automatically where it is appropriate to do so. "r" makes it go the next step and additionally generate ring tones where it is probably not appropriate to do so. ___ 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] Disconnect supervision question
Your line must provide disconnect supervision, which it sound like you say that it does, and you must configure that Zap trunk in asterisk for "kewl start" signaling (fxoks) which makes use of the disconnect supervision. On 8/18/05, James Fogg <[EMAIL PROTECTED]> wrote: > I'm trying to use Asterisk as a voice mail system and all is going well > except for one issue. When the outside caller drops the connection > Asterisk doesn't sense it. It keeps the line open for about 2 minutes > before dropping it. The FXO is attached to a key system (hybrid PBX). > There was a home-type digital answering machine previously on the same > extension and it was able to sense disconnect without any problems. > > The interface is a non-Digium single port FXO card (modem) with R13 & > R19 removed to mimic the Digium card. The software is the current > release of [EMAIL PROTECTED] from ISO format (self booting CD that does an > OS install and Asterisk compile automagically). > > My question is if Asterisk has settings to control disconnect sense, or > do I have to look at the key system (hybrid PBX) that it's attached to. > > -James > ___ > 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] Disconnect supervision question
I'm trying to use Asterisk as a voice mail system and all is going well except for one issue. When the outside caller drops the connection Asterisk doesn't sense it. It keeps the line open for about 2 minutes before dropping it. The FXO is attached to a key system (hybrid PBX). There was a home-type digital answering machine previously on the same extension and it was able to sense disconnect without any problems. The interface is a non-Digium single port FXO card (modem) with R13 & R19 removed to mimic the Digium card. The software is the current release of [EMAIL PROTECTED] from ISO format (self booting CD that does an OS install and Asterisk compile automagically). My question is if Asterisk has settings to control disconnect sense, or do I have to look at the key system (hybrid PBX) that it's attached to. -James ___ 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] Disconnect Supervision, SBC, and Adit 600
Can anyone help me with the term that SBC uses to refer to disconnect supervision? I have an Adit 600 channel bank which has helped improve the disconnect detection time down to about 8 seconds. This is still causing some issues in particular with call progress on we are having a few disconnects while calls are in session. I have talked both to some local phone contractors and SBC directly and no-one seems to know what I am talking about. The phone contractors new about the issue with other phone systems but didn't know there was a way to fix it and SBC reps seem to never have heard of disconnect sup. Also I have seen some references to loop start with call sup as kewlstart is there another name for this protocol? One of the local contractors thought that SBC automatically drops line voltage on remote hangup, in which case I need to know what signalling to program into the ADIT 600. I also have the option of going to groundstart signalling if this would fix the problem. -- Jonathan Moore Director of Technology Winfield Public Schools Office 620.221.5100 Fax 620.221.0508 Visit Winfield Public Schools at http://usd465.com - This mail sent through IMP: http://horde.org/imp/ ___ 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
[Asterisk-Users] Disconnect Supervision/CPC/CPD from Sprint?
Title: Message Has anyone on the list had any luck getting Sprint to enable Disconnect Supervision/CPC/CPD on plain analog lines? If so, what department and termonolgy did you use? Thanks, Marcus
