Re: [asterisk-users] Occasional call from asterisk
Thanks for the input. Long ago the CDR showed asterisk as the CLID but it
doesn't anymore so I am puzzled now how to even stop taking calls because my
CLID is now blank and I can't refuse any call with no CLID.
*WARNING[11002] chan_dahdi.c: CallerID returned with error on channel
'DAHDI/2-1'*
Here are some out of place messages I am getting in my logs but nothing out
of norm around the time I get Ghost calls though:
*WARNING[11002] chan_dahdi.c: CallerID returned with error on channel
'DAHDI/2-1'*
*NOTICE[12524] chan_dahdi.c: Got event 17 (Polarity Reversal)...*
*
*
*
DEBUG[12524] chan_dahdi.c: Ignore switch to REVERSED Polarity on channel 4,
state 6
DEBUG[12524] chan_dahdi.c: Ignoring Polarity switch to IDLE on channel 4,
state 6
*
Can someone shed light on these options as to what exactly they do:
hanguponpolarityswitch=yes
answeronpolarityswitch=yes
Hopefully some Asterisk guru can tell us more about what might be happening
as I see this as a situation that can be avoided or at least there should be
a workaround for this.
Regards,
On Mon, May 9, 2011 at 9:50 AM, Brian Henning bhenn...@pineinst.com wrote:
Hello Bruce,
I did not find a solution, only advice to lead me to think “huh, well
that’s annoying but we can deal with it.” I understand from my users,
though, that it’s *not* always the case that it’s a phantom call—sometimes
there really is someone calling.
Note that I haven’t tried what I’m about to suggest, but you might try
examining the CALLERID data before dialing the SIP extensions and, if it is
empty or contains “asterisk,” reset it to something like “not available.”
Cheers,
~Brian
*From:* Bruce B [mailto:bruceb...@gmail.com]
*Sent:* Friday, May 06, 2011 10:55 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Cc:* bhenn...@pineinst.com
*Subject:* Re: [asterisk-users] Occasional call from asterisk
Hi Brian,
Did you find a solution to your problem? or at least got a working
dial-plan for it? I have the same problem again as well and want to know
what to do with the dial-plan to off-set the effect at least since Telco
says it's not their issue.
Regards,
Bruce
On Thu, Apr 7, 2011 at 5:53 PM, Brian Henning bhenn...@pineinst.com
wrote:
Hi,
Now and then our SIP phones ring with asterisk showing as the caller-ID.
Upon picking up the receiver, there is about five seconds of silence and
then the channel is closed (hangup). Can anyone offer some insight?
Here's
relevant snippets from my extensions.conf and Master.csv log:
This line shows up in Master.csv:
,,1-NOANSWER,inbound,,DAHDI/1-1,SIP/505-0150,Dial,SIP/5
01SIP/502SIP/503SIP/504SIP/505SIP/506,10,tTgr,2011-04-07
21:37:05,2011-04-07 21:37:16,2011-04-07
21:37:21,16,5,ANSWERED,DOCUMENTATION,1302212225.444,
Here's [inbound] from extensions.conf:
[inbound]
exten = s,1,Answer
exten = s,n,Ringing
exten = s,n,Set(CALLERID(num),9${CALLERID(num)})
exten = s,n,Dial(SIP/504SIP/506,5,tTgr)
exten = s,n,Goto(1-${DIALSTATUS},1)
exten = 1-ANSWER,1,Hangup
exten =
_1-.,1,Dial(SIP/501SIP/502SIP/503SIP/504SIP/505SIP/506,10,tTgr)
exten = _1-.,n,Goto(2-${DIALSTATUS},1)
exten = 2-ANSWER,1,Hangup
exten = _2-.,1,Voicemail(499@default,u)
exten = _2-.,2,Hangup
The idea is that first 504 and 506 ring, then if neither of them answer,
everyone rings. Works great most of the time.
I have a hunch that maybe this happens if the inbound caller hangs up while
the first Dial() is ringing, but I would've expected to see the first Dial
(to 504 and 506) show up in the Master.csv log, and it's not there. (The
preceding line of the log is a call from almost an hour earlier). In that
case though I'd expect to see 1-CANCEL in the log instead. Perhaps if
the
caller happens to hang up right between the two Dial() commands?..
As an aside, the Set(CALLERID...) bit doesn't work. The idea was to
prepend
a 9 so that a SIP user could use the redial feature of the phone's call
log to return a missed call (automatically including the 9 for outside
line). Unfortunately the 9 does not get prepended.
Thanks in advance for any and all advice!
~Brian
--
Brian Henning, Software Engineer
/\Pine Research Instrumentation
//\\ 5908 Triangle Drive
///\\\ Raleigh, NC 27617
USA
||
||phone: 919.782.8320
fax: 919.782.8323
email: bhenn...@pineinst.com
--
--
_
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Re: [asterisk-users] Occasional call from asterisk
Hi Brian,
Did you find a solution to your problem? or at least got a working dial-plan
for it? I have the same problem again as well and want to know what to do
with the dial-plan to off-set the effect at least since Telco says it's not
their issue.
Regards,
Bruce
On Thu, Apr 7, 2011 at 5:53 PM, Brian Henning bhenn...@pineinst.com wrote:
Hi,
Now and then our SIP phones ring with asterisk showing as the caller-ID.
Upon picking up the receiver, there is about five seconds of silence and
then the channel is closed (hangup). Can anyone offer some insight?
Here's
relevant snippets from my extensions.conf and Master.csv log:
This line shows up in Master.csv:
,,1-NOANSWER,inbound,,DAHDI/1-1,SIP/505-0150,Dial,SIP/5
01SIP/502SIP/503SIP/504SIP/505SIP/506,10,tTgr,2011-04-07
21:37:05,2011-04-07 21:37:16,2011-04-07
21:37:21,16,5,ANSWERED,DOCUMENTATION,1302212225.444,
Here's [inbound] from extensions.conf:
[inbound]
exten = s,1,Answer
exten = s,n,Ringing
exten = s,n,Set(CALLERID(num),9${CALLERID(num)})
exten = s,n,Dial(SIP/504SIP/506,5,tTgr)
exten = s,n,Goto(1-${DIALSTATUS},1)
exten = 1-ANSWER,1,Hangup
exten =
_1-.,1,Dial(SIP/501SIP/502SIP/503SIP/504SIP/505SIP/506,10,tTgr)
exten = _1-.,n,Goto(2-${DIALSTATUS},1)
exten = 2-ANSWER,1,Hangup
exten = _2-.,1,Voicemail(499@default,u)
exten = _2-.,2,Hangup
The idea is that first 504 and 506 ring, then if neither of them answer,
everyone rings. Works great most of the time.
I have a hunch that maybe this happens if the inbound caller hangs up while
the first Dial() is ringing, but I would've expected to see the first Dial
(to 504 and 506) show up in the Master.csv log, and it's not there. (The
preceding line of the log is a call from almost an hour earlier). In that
case though I'd expect to see 1-CANCEL in the log instead. Perhaps if
the
caller happens to hang up right between the two Dial() commands?..
As an aside, the Set(CALLERID...) bit doesn't work. The idea was to
prepend
a 9 so that a SIP user could use the redial feature of the phone's call
log to return a missed call (automatically including the 9 for outside
line). Unfortunately the 9 does not get prepended.
Thanks in advance for any and all advice!
~Brian
--
Brian Henning, Software Engineer
/\Pine Research Instrumentation
//\\ 5908 Triangle Drive
///\\\ Raleigh, NC 27617
USA
||
||phone: 919.782.8320
fax: 919.782.8323
email: bhenn...@pineinst.com
--
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Re: [asterisk-users] Occasional call from asterisk
Telco always says it is not their issue.
This is all over google, did you even check? Did you check your options in
chan_dahdi.conf?
hanguponpolarityswitch=yes
I am not sure if that is your problem but it would be helpful to list the
things you have found, tested, and ruled out.
As for prepending a 9 for redial, I would say doing it in the [outbound]
dial context would be best practice.
For my installations, I have eliminated the need to Dial 9 for an outside
line That goes back to the key systems where 9 got you an outside line.
I have also eliminated the need to dial 1 as well. A good dialplan makes
these legacy, I still leave them there to avoid confusion.
For some clients that use TDM and VoIP, I may make 8 + number go over VoIP
and 9 + whatever go over TDM.
Default without the 8 or 9 is to go out over TDM or whatever the customer
wants, or TDM if they seem lost.
I don't give them too many decisions to make, just educate them on the
options programmed into the system.
Last thing, your dialplan looks too over engineered. How about this and
fixing your callerID syntax?
[inbound]
exten = s,1,Answer
exten = s,n,Set(CALLERID(num),9${CALLERID(num)})
exten = s,n,Dial(SIP/504SIP/506,5,tTgr)
exten = s,n,Dial(SIP/501SIP/502SIP/503SIP/504SIP/505SIP/506,10,tTgr)
exten = s,n,Voicemail(499@default,u)
exten = s,n,Hangup
Thanks,
Steve Totaro
On Fri, May 6, 2011 at 10:54 PM, Bruce B bruceb...@gmail.com wrote:
Hi Brian,
Did you find a solution to your problem? or at least got a working
dial-plan for it? I have the same problem again as well and want to know
what to do with the dial-plan to off-set the effect at least since Telco
says it's not their issue.
Regards,
Bruce
On Thu, Apr 7, 2011 at 5:53 PM, Brian Henning bhenn...@pineinst.comwrote:
Hi,
Now and then our SIP phones ring with asterisk showing as the caller-ID.
Upon picking up the receiver, there is about five seconds of silence and
then the channel is closed (hangup). Can anyone offer some insight?
Here's
relevant snippets from my extensions.conf and Master.csv log:
This line shows up in Master.csv:
,,1-NOANSWER,inbound,,DAHDI/1-1,SIP/505-0150,Dial,SIP/5
01SIP/502SIP/503SIP/504SIP/505SIP/506,10,tTgr,2011-04-07
21:37:05,2011-04-07 21:37:16,2011-04-07
21:37:21,16,5,ANSWERED,DOCUMENTATION,1302212225.444,
Here's [inbound] from extensions.conf:
[inbound]
exten = s,1,Answer
exten = s,n,Ringing
exten = s,n,Set(CALLERID(num),9${CALLERID(num)})
exten = s,n,Dial(SIP/504SIP/506,5,tTgr)
exten = s,n,Goto(1-${DIALSTATUS},1)
exten = 1-ANSWER,1,Hangup
exten =
_1-.,1,Dial(SIP/501SIP/502SIP/503SIP/504SIP/505SIP/506,10,tTgr)
exten = _1-.,n,Goto(2-${DIALSTATUS},1)
exten = 2-ANSWER,1,Hangup
exten = _2-.,1,Voicemail(499@default,u)
exten = _2-.,2,Hangup
The idea is that first 504 and 506 ring, then if neither of them answer,
everyone rings. Works great most of the time.
I have a hunch that maybe this happens if the inbound caller hangs up
while
the first Dial() is ringing, but I would've expected to see the first Dial
(to 504 and 506) show up in the Master.csv log, and it's not there. (The
preceding line of the log is a call from almost an hour earlier). In that
case though I'd expect to see 1-CANCEL in the log instead. Perhaps if
the
caller happens to hang up right between the two Dial() commands?..
As an aside, the Set(CALLERID...) bit doesn't work. The idea was to
prepend
a 9 so that a SIP user could use the redial feature of the phone's call
log to return a missed call (automatically including the 9 for outside
line). Unfortunately the 9 does not get prepended.
Thanks in advance for any and all advice!
~Brian
--
Brian Henning, Software Engineer
/\Pine Research Instrumentation
//\\ 5908 Triangle Drive
///\\\ Raleigh, NC 27617
USA
||
||phone: 919.782.8320
fax: 919.782.8323
email: bhenn...@pineinst.com
--
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Re: [asterisk-users] Occasional call from asterisk
Bruce B said: We experience exact same thing on DAHDI with Sangoma USB FXO device on short circuited lines. Phantom calls are actually due to a short in the lines that happen occasionally. -Bruce Also, Warren Selby said: I've seen this on cases where a phantom call comes in on a DAHDI channel - these calls were the results of faulty wiring on the part of the telco. Check your logs for any errors on your DAHDI channels around the time of the ghost calls. It could also be a case of someone calls in and then hangs up before the call is actually passed to asterisk, and the telco is just slow to hangup the call. H. I do see this in the /var/log/asterisk/messages log: [Apr 5 00:05:36] NOTICE[9579] chan_dahdi.c: Got event 4 (Alarm)... [Apr 5 00:05:38] NOTICE[9579] chan_dahdi.c: Got event 5 (No more alarm)... [Apr 5 00:05:38] NOTICE[9579] chan_dahdi.c: Got event 17 (Polarity Reversal)... [Apr 5 00:05:41] WARNING[2400] chan_dahdi.c: Detected alarm on channel 3: Red Alarm [Apr 5 00:05:42] NOTICE[2400] chan_dahdi.c: Alarm cleared on channel 3 [Apr 5 00:05:51] WARNING[2400] chan_dahdi.c: Detected alarm on channel 2: Red Alarm [Apr 5 00:05:53] NOTICE[2400] chan_dahdi.c: Alarm cleared on channel 2 .and it appears to coincide directly with an 'asterisk' entry in my SIP phone's missed call log. Our wiring is sketchy; this is known at our facility. Some years ago a backhoe severed the entire trunk and the repair work was of questionable quality. Also our service entry point / punch-down area is a rat's nest (one building and service is shared by three companies). I guess I can chalk this behavior up to the wiring. Thanks for the input! Cheers, ~Brian -- _ -- 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] Occasional call from asterisk
_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Brian Henning Sent: Monday, April 11, 2011 8:47 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Occasional call from asterisk Bruce B said: We experience exact same thing on DAHDI with Sangoma USB FXO device on short circuited lines. Phantom calls are actually due to a short in the lines that happen occasionally. -Bruce Also, Warren Selby said: I've seen this on cases where a phantom call comes in on a DAHDI channel - these calls were the results of faulty wiring on the part of the telco. Check your logs for any errors on your DAHDI channels around the time of the ghost calls. It could also be a case of someone calls in and then hangs up before the call is actually passed to asterisk, and the telco is just slow to hangup the call. H. I do see this in the /var/log/asterisk/messages log: [Apr 5 00:05:36] NOTICE[9579] chan_dahdi.c: Got event 4 (Alarm)... [Apr 5 00:05:38] NOTICE[9579] chan_dahdi.c: Got event 5 (No more alarm)... [Apr 5 00:05:38] NOTICE[9579] chan_dahdi.c: Got event 17 (Polarity Reversal)... [Apr 5 00:05:41] WARNING[2400] chan_dahdi.c: Detected alarm on channel 3: Red Alarm [Apr 5 00:05:42] NOTICE[2400] chan_dahdi.c: Alarm cleared on channel 3 [Apr 5 00:05:51] WARNING[2400] chan_dahdi.c: Detected alarm on channel 2: Red Alarm [Apr 5 00:05:53] NOTICE[2400] chan_dahdi.c: Alarm cleared on channel 2 .and it appears to coincide directly with an 'asterisk' entry in my SIP phone's missed call log. Our wiring is sketchy; this is known at our facility. Some years ago a backhoe severed the entire trunk and the repair work was of questionable quality. Also our service entry point / punch-down area is a rat's nest (one building and service is shared by three companies). I guess I can chalk this behavior up to the wiring. Thanks for the input! Cheers, ~Brian [Danny Nicholas] I'll add another vote to shoot the phone company - our wiring goes to heck whenever it rains and we can expect a few 'phantom calls from Asterisk. -- _ -- 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] Occasional call from asterisk
I wonder if you can test to see if this happens if you had an analogue phone set connected. And if it doesn't then I am wondering why Asterisk or Sangoma card is so sensitive and maybe the sensor can be set a bit higher so these calls don't end-up ringing like they don't if an analogue phone set was connected to the line (at least that was my case). -Bruce On Mon, Apr 11, 2011 at 9:52 AM, Danny Nicholas da...@debsinc.com wrote: -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Brian Henning *Sent:* Monday, April 11, 2011 8:47 AM *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion' *Subject:* Re: [asterisk-users] Occasional call from asterisk Bruce B said: We experience exact same thing on DAHDI with Sangoma USB FXO device on short circuited lines. Phantom calls are actually due to a short in the lines that happen occasionally. -Bruce Also, Warren Selby said: I've seen this on cases where a phantom call comes in on a DAHDI channel - these calls were the results of faulty wiring on the part of the telco. Check your logs for any errors on your DAHDI channels around the time of the ghost calls. It could also be a case of someone calls in and then hangs up before the call is actually passed to asterisk, and the telco is just slow to hangup the call. H. I do see this in the /var/log/asterisk/messages log: [Apr 5 00:05:36] NOTICE[9579] chan_dahdi.c: Got event 4 (Alarm)... [Apr 5 00:05:38] NOTICE[9579] chan_dahdi.c: Got event 5 (No more alarm)... [Apr 5 00:05:38] NOTICE[9579] chan_dahdi.c: Got event 17 (Polarity Reversal)... [Apr 5 00:05:41] WARNING[2400] chan_dahdi.c: Detected alarm on channel 3: Red Alarm [Apr 5 00:05:42] NOTICE[2400] chan_dahdi.c: Alarm cleared on channel 3 [Apr 5 00:05:51] WARNING[2400] chan_dahdi.c: Detected alarm on channel 2: Red Alarm [Apr 5 00:05:53] NOTICE[2400] chan_dahdi.c: Alarm cleared on channel 2 …and it appears to coincide directly with an ‘asterisk’ entry in my SIP phone’s missed call log. Our wiring is sketchy; this is known at our facility. Some years ago a backhoe severed the entire trunk and the repair work was of questionable quality. Also our service entry point / punch-down area is a rat’s nest (one building and service is shared by three companies). I guess I can chalk this behavior up to the wiring. Thanks for the input! Cheers, ~Brian *[Danny Nicholas] * *I’ll add another vote to “shoot the phone company” – our wiring goes to heck whenever it rains and we can expect a few ‘phantom calls” from Asterisk.* -- _ -- 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] Occasional call from asterisk
On Mon, Apr 11, 2011 at 8:47 AM, Brian Henning bhenn...@pineinst.comwrote: H. I do see this in the /var/log/asterisk/messages log: [Apr 5 00:05:36] NOTICE[9579] chan_dahdi.c: Got event 4 (Alarm)... [Apr 5 00:05:38] NOTICE[9579] chan_dahdi.c: Got event 5 (No more alarm)... [Apr 5 00:05:38] NOTICE[9579] chan_dahdi.c: Got event 17 (Polarity Reversal)... I used to see this a LOT with an old cheapo X100P card. It always seemed to happen around the same time of day too, about 9:00pm. Haven't had the problem since I switched to a real TDM410 card. -- Chris -- _ -- 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] Occasional call from asterisk
Hi,
Now and then our SIP phones ring with asterisk showing as the caller-ID.
Upon picking up the receiver, there is about five seconds of silence and
then the channel is closed (hangup). Can anyone offer some insight? Here's
relevant snippets from my extensions.conf and Master.csv log:
This line shows up in Master.csv:
,,1-NOANSWER,inbound,,DAHDI/1-1,SIP/505-0150,Dial,SIP/5
01SIP/502SIP/503SIP/504SIP/505SIP/506,10,tTgr,2011-04-07
21:37:05,2011-04-07 21:37:16,2011-04-07
21:37:21,16,5,ANSWERED,DOCUMENTATION,1302212225.444,
Here's [inbound] from extensions.conf:
[inbound]
exten = s,1,Answer
exten = s,n,Ringing
exten = s,n,Set(CALLERID(num),9${CALLERID(num)})
exten = s,n,Dial(SIP/504SIP/506,5,tTgr)
exten = s,n,Goto(1-${DIALSTATUS},1)
exten = 1-ANSWER,1,Hangup
exten =
_1-.,1,Dial(SIP/501SIP/502SIP/503SIP/504SIP/505SIP/506,10,tTgr)
exten = _1-.,n,Goto(2-${DIALSTATUS},1)
exten = 2-ANSWER,1,Hangup
exten = _2-.,1,Voicemail(499@default,u)
exten = _2-.,2,Hangup
The idea is that first 504 and 506 ring, then if neither of them answer,
everyone rings. Works great most of the time.
I have a hunch that maybe this happens if the inbound caller hangs up while
the first Dial() is ringing, but I would've expected to see the first Dial
(to 504 and 506) show up in the Master.csv log, and it's not there. (The
preceding line of the log is a call from almost an hour earlier). In that
case though I'd expect to see 1-CANCEL in the log instead. Perhaps if the
caller happens to hang up right between the two Dial() commands?..
As an aside, the Set(CALLERID...) bit doesn't work. The idea was to prepend
a 9 so that a SIP user could use the redial feature of the phone's call
log to return a missed call (automatically including the 9 for outside
line). Unfortunately the 9 does not get prepended.
Thanks in advance for any and all advice!
~Brian
--
Brian Henning, Software Engineer
/\Pine Research Instrumentation
//\\ 5908 Triangle Drive
///\\\ Raleigh, NC 27617
USA
||
||phone: 919.782.8320
fax: 919.782.8323
email: bhenn...@pineinst.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
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Re: [asterisk-users] Occasional call from asterisk
We were getting a lot of those. We installed IPTables with blocking of
everything outside of North America and they all but vanished.
No direct evidence, but a pretty good empirical guess that they were related
to hackers trying to get paths to the US.
CF
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Brian Henning
Sent: Thursday, April 07, 2011 4:54 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] Occasional call from asterisk
Hi,
Now and then our SIP phones ring with asterisk showing as the caller-ID.
Upon picking up the receiver, there is about five seconds of silence and
then the channel is closed (hangup). Can anyone offer some insight? Here's
relevant snippets from my extensions.conf and Master.csv log:
This line shows up in Master.csv:
,,1-NOANSWER,inbound,,DAHDI/1-1,SIP/505-0150,Dial,SIP/5
01SIP/502SIP/503SIP/504SIP/505SIP/506,10,tTgr,2011-04-07
21:37:05,2011-04-07 21:37:16,2011-04-07
21:37:21,16,5,ANSWERED,DOCUMENTATION,1302212225.444,
Here's [inbound] from extensions.conf:
[inbound]
exten = s,1,Answer
exten = s,n,Ringing
exten = s,n,Set(CALLERID(num),9${CALLERID(num)})
exten = s,n,Dial(SIP/504SIP/506,5,tTgr)
exten = s,n,Goto(1-${DIALSTATUS},1)
exten = 1-ANSWER,1,Hangup
exten =
_1-.,1,Dial(SIP/501SIP/502SIP/503SIP/504SIP/505SIP/506,10,tTgr)
exten = _1-.,n,Goto(2-${DIALSTATUS},1)
exten = 2-ANSWER,1,Hangup
exten = _2-.,1,Voicemail(499@default,u)
exten = _2-.,2,Hangup
The idea is that first 504 and 506 ring, then if neither of them answer,
everyone rings. Works great most of the time.
I have a hunch that maybe this happens if the inbound caller hangs up while
the first Dial() is ringing, but I would've expected to see the first Dial
(to 504 and 506) show up in the Master.csv log, and it's not there. (The
preceding line of the log is a call from almost an hour earlier). In that
case though I'd expect to see 1-CANCEL in the log instead. Perhaps if the
caller happens to hang up right between the two Dial() commands?..
As an aside, the Set(CALLERID...) bit doesn't work. The idea was to prepend
a 9 so that a SIP user could use the redial feature of the phone's call
log to return a missed call (automatically including the 9 for outside
line). Unfortunately the 9 does not get prepended.
Thanks in advance for any and all advice!
~Brian
--
Brian Henning, Software Engineer
/\Pine Research Instrumentation
//\\ 5908 Triangle Drive
///\\\ Raleigh, NC 27617
USA
||
||phone: 919.782.8320
fax: 919.782.8323
email: bhenn...@pineinst.com
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Re: [asterisk-users] Occasional call from asterisk
We experience exact same thing on DAHDI with Sangoma USB FXO device on short
circuited lines. Phantom calls are actually due to a short in the lines that
happen occasionally.
-Bruce
On Thu, Apr 7, 2011 at 7:16 PM, Warren Selby wcse...@selbytech.com wrote:
On Thu, Apr 7, 2011 at 4:53 PM, Brian Henning bhenn...@pineinst.comwrote:
Hi,
Now and then our SIP phones ring with asterisk showing as the caller-ID.
Upon picking up the receiver, there is about five seconds of silence and
then the channel is closed (hangup). Can anyone offer some insight?
Here's
relevant snippets from my extensions.conf and Master.csv log:
snip
I've seen this on cases where a phantom call comes in on a DAHDI channel
- these calls were the results of faulty wiring on the part of the telco.
Check your logs for any errors on your DAHDI channels around the time of the
ghost calls.
It could also be a case of someone calls in and then hangs up before the
call is actually passed to asterisk, and the telco is just slow to hangup
the call.
As an aside, the Set(CALLERID...) bit doesn't work. The idea was to
prepend
a 9 so that a SIP user could use the redial feature of the phone's call
log to return a missed call (automatically including the 9 for outside
line). Unfortunately the 9 does not get prepended.
Your Set() syntax is wrong. Try this:
exten = s,n,Set(CALLERID(num)=9${CALLERID(num)})
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Thanks,
--Warren Selby, dCAP
http://www.selbytech.com
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