Re: [asterisk-users] Phantom Calls
Lee Jenkins wrote on 6/19/07 9:56 AM: Vadim Berezniker wrote: Enable verbose logging for the asterisk log Set verbose level to 4 Review the log file for anything that looks like a phantom call. There should be enough information to get some idea of why this is happening. Thanks, I have done this yesterday by setting up putty to log to a file, but the customer employees have inadvertently shut it down on a couple of a occasions :) Hopefully it will be running when this happens again so I can try to track down the problem. You should be able to tell it to log to a file in addition to the console in logger.conf. Something like: full = notice,warning,error,verbose Then it should show up in /var/log/asterisk/full and you wouldn't need to keep a session open to the console to see it, just go back and look at the file later. -- Dave Miller http://www.justdave.net/ System Administrator, Mozilla Corporation http://www.mozilla.com/ Project Leader, Bugzilla Bug Tracking System http://www.bugzilla.org/ ___ --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] Phantom Calls
Dave Miller wrote: Lee Jenkins wrote on 6/19/07 9:56 AM: Vadim Berezniker wrote: Enable verbose logging for the asterisk log Set verbose level to 4 Review the log file for anything that looks like a phantom call. There should be enough information to get some idea of why this is happening. Thanks, I have done this yesterday by setting up putty to log to a file, but the customer employees have inadvertently shut it down on a couple of a occasions :) Hopefully it will be running when this happens again so I can try to track down the problem. You should be able to tell it to log to a file in addition to the console in logger.conf. Something like: full = notice,warning,error,verbose Then it should show up in /var/log/asterisk/full and you wouldn't need to keep a session open to the console to see it, just go back and look at the file later. Nice tip, Dave. Thanks, -- Warm Regards, Lee ___ --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] Phantom Calls
Vadim Berezniker wrote: Enable verbose logging for the asterisk log Set verbose level to 4 Review the log file for anything that looks like a phantom call. There should be enough information to get some idea of why this is happening. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lee Jenkins Sent: Monday, June 18, 2007 1:56 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Phantom Calls Stephen Bosch wrote: Lee Jenkins wrote: I have a client that is having problems with phantom calls. I have not been able to see it happen myself, but they say when it happens, the display on the phone (polycom 301's) says Device is calling, but when they answer the phone, there is only silence and then they hang back up and it sometimes rings again. I've been trying to track this down for a couple of weeks now with no success yet. If anyone can lend a suggestion or a pointer to look for, I would greatly appreciate it. I've tried using WaitForRing() in case it is bad signaling coming from the phone company. But that has not helped. Below is the CLI output when this issue happened. As you can see, I am using WaitForRing() to discourage phantom calls. Every time this has happened, there appears to be an error getting caller ID. I'm thinking that if I insert a Wait(1/2) before Answer, that may resolve the problems with Caller ID as it looks like Asterisk is not waiting long enough for the CID to come in. Whether or not that will fix the problem with phantom calls remains to be seen after I make the changes. Also notice, the line: localhost*CLI -- Got SIP response 400 Bad Request back from 192.168.1.216. What does 400 Bad Request usually mean for sip? Generic message or something that would provide a clue? localhost*CLI -- Starting simple switch on 'Zap/3-1' localhost*CLI Jun 21 10:44:48 NOTICE[11257]: callerid.c:325 callerid_feed: Caller*ID failed checksum localhost*CLI Jun 21 10:44:51 NOTICE[11257]: chan_zap.c:6233 ss_thread: Got event 18 (Ring Begin)... localhost*CLI Jun 21 10:44:53 NOTICE[11257]: chan_zap.c:6233 ss_thread: Got event 2 (Ring/Answered)... -- Executing WaitForRing(Zap/3-1, 1) in new stack localhost*CLI -- Got a ring after the timeout -- Executing Answer(Zap/3-1, ) in new stack -- Executing Ringing(Zap/3-1, ) in new stack -- Executing SetMusicOnHold(Zap/3-1, default) in new stack -- Executing Goto(Zap/3-1, check_time|s|1) in new stack -- Goto (check_time,s,1) -- Executing Set(Zap/3-1, FAIL_MENU=daytime|TIMEOUT_MENU=daytime) in new stack -- Executing GotoIfTime(Zap/3-1, 08:30-17:00|mon-fri|*|*|?daytime|s|1) in new stack -- Goto (daytime,s,1) -- Executing Set(Zap/3-1, TIMEOUT(response)=1) in new stack -- Response timeout set to 1 -- Executing Dial(Zap/3-1, SIP/114SIP/115SIP/116SIP/117|20|tr) in new stack -- Called 114 -- Called 115 -- Called 116 -- Called 117 localhost*CLI -- SIP/117-0a0718d8 is ringing localhost*CLI -- SIP/116-0a06c398 is ringing localhost*CLI -- SIP/115-0a057678 is ringing localhost*CLI -- SIP/114-0a066c58 is ringing localhost*CLI -- Nobody picked up in 2 ms -- Executing BackGround(Zap/3-1, custom/no-answer) in new stack -- Playing 'custom/no-answer' (language 'en') localhost*CLI -- Got SIP response 400 Bad Request back from 192.168.1.216 localhost*CLI -- Timeout on Zap/3-1 == CDR updated on Zap/3-1 -- Executing Goto(Zap/3-1, daytime|s|1) in new stack -- Goto (daytime,s,1) -- Executing Set(Zap/3-1, TIMEOUT(response)=1) in new stack -- Response timeout set to 1 -- Executing Dial(Zap/3-1, SIP/114SIP/115SIP/116SIP/117|20|tr) in new stack -- Called 114 -- Called 115 -- Called 116 -- Called 117 localhost*CLI -- SIP/115-0a076e18 is ringing localhost*CLI -- SIP/116-0a07c358 is ringing localhost*CLI -- SIP/114-0a06c398 is ringing localhost*CLI -- SIP/117-0a081898 is ringing localhost*CLI -- Nobody picked up in 2 ms -- Executing BackGround(Zap/3-1, custom/no-answer) in new stack -- Playing 'custom/no-answer' (language 'en') localhost*CLI -- Got SIP response 400 Bad Request back from 192.168.1.216 localhost*CLI -- Timeout on Zap/3-1 == CDR updated on Zap/3-1 -- Executing Goto(Zap/3-1, daytime|s|1) in new stack -- Goto (daytime,s,1) -- Executing Set(Zap/3-1, TIMEOUT(response)=1) in new stack -- Response timeout set to 1 -- Executing Dial(Zap/3-1, SIP/114SIP/115SIP/116SIP/117|20|tr) in new stack -- Called 114 -- Called 115 -- Called 116 -- Called 117 localhost*CLI -- SIP/116-0a07c358 is ringing localhost*CLI -- SIP/117-0a0718d8 is ringing localhost*CLI -- SIP/115-0a057678 is ringing localhost*CLI -- SIP/114-0a066c58 is ringing localhost*CLI == Spawn extension
Re: [asterisk-users] Phantom Calls
Enable verbose logging for the asterisk log Set verbose level to 4 Review the log file for anything that looks like a phantom call. There should be enough information to get some idea of why this is happening. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lee Jenkins Sent: Monday, June 18, 2007 1:56 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Phantom Calls Stephen Bosch wrote: Lee Jenkins wrote: I have a client that is having problems with phantom calls. I have not been able to see it happen myself, but they say when it happens, the display on the phone (polycom 301's) says Device is calling, but when they answer the phone, there is only silence and then they hang back up and it sometimes rings again. I've been trying to track this down for a couple of weeks now with no success yet. If anyone can lend a suggestion or a pointer to look for, I would greatly appreciate it. I've tried using WaitForRing() in case it is bad signaling coming from the phone company. But that has not helped. ___ --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] Phantom Calls
Vadim Berezniker wrote: Enable verbose logging for the asterisk log Set verbose level to 4 Review the log file for anything that looks like a phantom call. There should be enough information to get some idea of why this is happening. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lee Jenkins Sent: Monday, June 18, 2007 1:56 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Phantom Calls Stephen Bosch wrote: Lee Jenkins wrote: I have a client that is having problems with phantom calls. I have not been able to see it happen myself, but they say when it happens, the display on the phone (polycom 301's) says Device is calling, but when they answer the phone, there is only silence and then they hang back up and it sometimes rings again. I've been trying to track this down for a couple of weeks now with no success yet. If anyone can lend a suggestion or a pointer to look for, I would greatly appreciate it. I've tried using WaitForRing() in case it is bad signaling coming from the phone company. But that has not helped. Thanks, I have done this yesterday by setting up putty to log to a file, but the customer employees have inadvertently shut it down on a couple of a occasions :) Hopefully it will be running when this happens again so I can try to track down the problem. This one is a sticky situation. This particular installation is for a friend of mine and his company. He decided to get a system from me instead of another friend of his that sells Panasonic or Avaya systems. -- Warm Regards, Lee ___ --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] Phantom Calls
Hi all, I have a client that is having problems with phantom calls. I have not been able to see it happen myself, but they say when it happens, the display on the phone (polycom 301's) says Device is calling, but when they answer the phone, there is only silence and then they hang back up and it sometimes rings again. I've been trying to track this down for a couple of weeks now with no success yet. If anyone can lend a suggestion or a pointer to look for, I would greatly appreciate it. I've tried using WaitForRing() in case it is bad signaling coming from the phone company. But that has not helped. This setup has: Asterisk 1.2.17 Zaptel (whatever was distributed same time as Asterisk 1.2.17) CentOS 4.4 Polycom 301's throughout Sangoma A200 with 2 ports connected to PSTN. Thanks for any help. -- Warm Regards, Lee ___ --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] Phantom Calls
Lee Jenkins wrote: I have a client that is having problems with phantom calls. I have not been able to see it happen myself, but they say when it happens, the display on the phone (polycom 301's) says Device is calling, but when they answer the phone, there is only silence and then they hang back up and it sometimes rings again. I've been trying to track this down for a couple of weeks now with no success yet. If anyone can lend a suggestion or a pointer to look for, I would greatly appreciate it. I've tried using WaitForRing() in case it is bad signaling coming from the phone company. But that has not helped. I highly doubt this is the cause. I suspect a dialplan error. Ask your client if they are doing anything else when this happens, like making a fax call ;) This setup has: Asterisk 1.2.17 Zaptel (whatever was distributed same time as Asterisk 1.2.17) CentOS 4.4 Polycom 301's throughout Sangoma A200 with 2 ports connected to PSTN. Thanks for any help. We kinda need to see your dialplan to provide any useful help. -Stephen- ___ --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] Phantom Calls
Stephen Bosch wrote:
Lee Jenkins wrote:
I have a client that is having problems with phantom calls. I have not
been able to see it happen myself, but they say when it happens, the
display on the phone (polycom 301's) says Device is calling, but when
they answer the phone, there is only silence and then they hang back up
and it sometimes rings again.
I've been trying to track this down for a couple of weeks now with no
success yet. If anyone can lend a suggestion or a pointer to look for,
I would greatly appreciate it.
I've tried using WaitForRing() in case it is bad signaling coming from
the phone company. But that has not helped.
I highly doubt this is the cause. I suspect a dialplan error.
Ask your client if they are doing anything else when this happens, like
making a fax call ;)
This setup has:
Asterisk 1.2.17
Zaptel (whatever was distributed same time as Asterisk 1.2.17)
CentOS 4.4
Polycom 301's throughout
Sangoma A200 with 2 ports connected to PSTN.
Thanks for any help.
We kinda need to see your dialplan to provide any useful help.
-Stephen-
Thanks for responding, Stephen.
The client has a fax line, but it is separate line from the Asterisk
box. They have 3 lines coming in. 1 goes directly to fax machine and 2
go to Asterisk box.
I've searched the archives and phantom ringing comes up a few times
without any real resolutions that I can see.
The only thing very different about this installation is that the
customer has no intermediate IVR. They want the phones to ring directly
to a group and if no answer, then go to a mini IVR that asks if they
would like to hold longer or leave a message.
The device calling on CallerID when this happens is the default
CallerID set in sip.conf Device callernum which was never changed.
That was changed to a correct value. It appears as though the system is
calling itself.
extensions.conf:
[incoming]
exten=s,1,WaitForRing(5)
exten=s,n,Answer()
exten=s,n,Ringing()
exten=s,n,SetMusicOnHold(default)
exten=s,n,Wait(1)
exten=s,n,Goto(check_time,s,1)
[check_time]
exten=s,1,Answer()
exten=s,2,GotoIfTime(08:30-17:00,mon-fri,*,*,?daytime,s,1)
exten=s,3,Goto(after_hours,s,1)
[daytime]
exten=s,1,Answer()
exten=s,2,Set(TIMEOUT(response)=1)
exten=s,3,Dial(${GRP_All},20,tr)
exten=s,4,Background(custom/no-answer)
exten=1,1,Set(loops=0)
exten=1,2,Wait(0.5)
exten=1,3,Goto(ring-all-with-moh,s,1)
exten=2,1,Voicemail(b116)
exten=2,2,Hangup()
exten=5,1,Voicemail(b116)
exten=5,2,Hangup()
exten=555,1,VoicemailMain()
exten=i,1,Background(invalid)
exten=i,2,Goto(ring-all-with-moh,s,1)
exten=t,1,Set(loops=0)
exten=t,2,Wait(0.5)
exten=t,3,Goto(ring-all-with-moh,s,1)
[ring-all-with-moh]
exten=s,1,Answer()
exten=s,2,Noop(Loops are: ${loops})
exten=s,3,Macro(DialExtenNoVM,116|60|tm)
exten=s,4,Set(loops=$[${loops}+1])
exten=s,5,GotoIf($[${loops} = 2 ]?timeout-from-loop,s,1)
exten=s,6,Background(custom/no-answer)
exten=1,1,Goto(ring-all-with-moh,s,1)
exten=2,1,Voicemail(b116)
exten=2,2,Hangup()
exten=i,1,Playback(invalid)
exten=i,2,Goto(ring-all-with-moh,s,1)
exten=t,1,Goto(ring-all-with-moh,s,1)
sip.conf:
[general]
allowexternalinvites=yes
allowguest=no
autocreatepeer=no
autodomain=no
bindaddr=0.0.0.0
callerid=device callernum
canreinvite=no
checkmwi=30
compactheaders=no
context=incoming
defaultexpirey=120
dtmfmode=rfc2833
dumphistory=no
externrefresh=30
ignoreregexpire=no
insecure=no
maxexpirey=3600
musicclass=default
nat=no
notifyringing=yes
pedantic=no
progressinband=never
promiscredir=no
qualify=no
recordhistory=no
registerattempts=30
registertimeout=30
relaxdtmf=no
rtautoclear=no
rtcachefriends=no
rtpholdtimeout=600
rtpkeepalive=0
rtptimeout=3600
rtupdate=yes
sendrpid=no
sipdebug=no
srvlookup=no
tos=none
trustrpid=no
useclientcode=no
usereqphone=no
callevents=no
disallow=all
allow=ulaw
[116]
context=super-user
type=friend
canreinvite=no
dtmfmode=rfc2833
callerid=Barbara 116
nat=no
port=5060
qualify=no
secret=xxx
host=dynamic
[EMAIL PROTECTED]
disallow=all
allow=ulaw
[117]
context=super-user
type=friend
canreinvite=no
dtmfmode=rfc2833
callerid=Bill 117
nat=no
port=5060
qualify=no
secret=xxx
host=dynamic
[EMAIL PROTECTED]
disallow=all
allow=ulaw
[115]
context=super-user
type=friend
canreinvite=no
dtmfmode=rfc2833
callerid=George 115
nat=no
port=5060
qualify=no
secret=xxx
host=dynamic
[EMAIL PROTECTED]
disallow=all
allow=ulaw
[114]
context=super-user
type=friend
canreinvite=no
dtmfmode=rfc2833
callerid=Jack 114
nat=no
port=5060
qualify=no
secret=xxx
host=dynamic
[EMAIL PROTECTED]
disallow=all
allow=ulaw
--
Warm Regards,
Lee
___
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Re: [asterisk-users] Phantom Calls
We were having phantom calls as well. In our case, we had 2 pots line
running in our sangoma card, and when you dial out, would would wait for
whomever to pickup. If you gave up waiting an hung the phone up (we also
had 2 normal phones plugged into fxs ports), it wouldn't immediately
receive the hangup signal. The call would connect, then asterisk would
turn around and try to call us back. If the other side hungup because
they just heard dead error, then when you'd repickup your call, it would
also be dead air.
Not sure if this is the same case as yours, but ours was odd as well.
Rob
Lee Jenkins wrote:
Stephen Bosch wrote:
Lee Jenkins wrote:
I have a client that is having problems with phantom calls. I have not
been able to see it happen myself, but they say when it happens, the
display on the phone (polycom 301's) says Device is calling, but when
they answer the phone, there is only silence and then they hang back up
and it sometimes rings again.
I've been trying to track this down for a couple of weeks now with no
success yet. If anyone can lend a suggestion or a pointer to look for,
I would greatly appreciate it.
I've tried using WaitForRing() in case it is bad signaling coming from
the phone company. But that has not helped.
I highly doubt this is the cause. I suspect a dialplan error.
Ask your client if they are doing anything else when this happens, like
making a fax call ;)
This setup has:
Asterisk 1.2.17
Zaptel (whatever was distributed same time as Asterisk 1.2.17)
CentOS 4.4
Polycom 301's throughout
Sangoma A200 with 2 ports connected to PSTN.
Thanks for any help.
We kinda need to see your dialplan to provide any useful help.
-Stephen-
Thanks for responding, Stephen.
The client has a fax line, but it is separate line from the Asterisk
box. They have 3 lines coming in. 1 goes directly to fax machine and 2
go to Asterisk box.
I've searched the archives and phantom ringing comes up a few times
without any real resolutions that I can see.
The only thing very different about this installation is that the
customer has no intermediate IVR. They want the phones to ring directly
to a group and if no answer, then go to a mini IVR that asks if they
would like to hold longer or leave a message.
The device calling on CallerID when this happens is the default
CallerID set in sip.conf Device callernum which was never changed.
That was changed to a correct value. It appears as though the system is
calling itself.
extensions.conf:
[incoming]
exten=s,1,WaitForRing(5)
exten=s,n,Answer()
exten=s,n,Ringing()
exten=s,n,SetMusicOnHold(default)
exten=s,n,Wait(1)
exten=s,n,Goto(check_time,s,1)
[check_time]
exten=s,1,Answer()
exten=s,2,GotoIfTime(08:30-17:00,mon-fri,*,*,?daytime,s,1)
exten=s,3,Goto(after_hours,s,1)
[daytime]
exten=s,1,Answer()
exten=s,2,Set(TIMEOUT(response)=1)
exten=s,3,Dial(${GRP_All},20,tr)
exten=s,4,Background(custom/no-answer)
exten=1,1,Set(loops=0)
exten=1,2,Wait(0.5)
exten=1,3,Goto(ring-all-with-moh,s,1)
exten=2,1,Voicemail(b116)
exten=2,2,Hangup()
exten=5,1,Voicemail(b116)
exten=5,2,Hangup()
exten=555,1,VoicemailMain()
exten=i,1,Background(invalid)
exten=i,2,Goto(ring-all-with-moh,s,1)
exten=t,1,Set(loops=0)
exten=t,2,Wait(0.5)
exten=t,3,Goto(ring-all-with-moh,s,1)
[ring-all-with-moh]
exten=s,1,Answer()
exten=s,2,Noop(Loops are: ${loops})
exten=s,3,Macro(DialExtenNoVM,116|60|tm)
exten=s,4,Set(loops=$[${loops}+1])
exten=s,5,GotoIf($[${loops} = 2 ]?timeout-from-loop,s,1)
exten=s,6,Background(custom/no-answer)
exten=1,1,Goto(ring-all-with-moh,s,1)
exten=2,1,Voicemail(b116)
exten=2,2,Hangup()
exten=i,1,Playback(invalid)
exten=i,2,Goto(ring-all-with-moh,s,1)
exten=t,1,Goto(ring-all-with-moh,s,1)
sip.conf:
[general]
allowexternalinvites=yes
allowguest=no
autocreatepeer=no
autodomain=no
bindaddr=0.0.0.0
callerid=device callernum
canreinvite=no
checkmwi=30
compactheaders=no
context=incoming
defaultexpirey=120
dtmfmode=rfc2833
dumphistory=no
externrefresh=30
ignoreregexpire=no
insecure=no
maxexpirey=3600
musicclass=default
nat=no
notifyringing=yes
pedantic=no
progressinband=never
promiscredir=no
qualify=no
recordhistory=no
registerattempts=30
registertimeout=30
relaxdtmf=no
rtautoclear=no
rtcachefriends=no
rtpholdtimeout=600
rtpkeepalive=0
rtptimeout=3600
rtupdate=yes
sendrpid=no
sipdebug=no
srvlookup=no
tos=none
trustrpid=no
useclientcode=no
usereqphone=no
callevents=no
disallow=all
allow=ulaw
[116]
context=super-user
type=friend
canreinvite=no
dtmfmode=rfc2833
callerid=Barbara 116
nat=no
port=5060
qualify=no
secret=xxx
host=dynamic
[EMAIL PROTECTED]
disallow=all
allow=ulaw
[117]
context=super-user
type=friend
canreinvite=no
dtmfmode=rfc2833
callerid=Bill 117
nat=no
port=5060
qualify=no
secret=xxx
host=dynamic
[EMAIL PROTECTED]
disallow=all
allow=ulaw
[115]
Re: [asterisk-users] Phantom Calls
I too have seen what Rob is saying.. on a Sangoma card. It was an easy fix
in the config, but I don't remember what it was.. but basically it was stray
voltage.
On 6/18/07, Rob Schall [EMAIL PROTECTED] wrote:
We were having phantom calls as well. In our case, we had 2 pots line
running in our sangoma card, and when you dial out, would would wait for
whomever to pickup. If you gave up waiting an hung the phone up (we also had
2 normal phones plugged into fxs ports), it wouldn't immediately receive the
hangup signal. The call would connect, then asterisk would turn around and
try to call us back. If the other side hungup because they just heard dead
error, then when you'd repickup your call, it would also be dead air.
Not sure if this is the same case as yours, but ours was odd as well.
Rob
Lee Jenkins wrote:
Stephen Bosch wrote:
Lee Jenkins wrote:
I have a client that is having problems with phantom calls. I have not
been able to see it happen myself, but they say when it happens, the
display on the phone (polycom 301's) says Device is calling, but when
they answer the phone, there is only silence and then they hang back up
and it sometimes rings again.
I've been trying to track this down for a couple of weeks now with no
success yet. If anyone can lend a suggestion or a pointer to look for,
I would greatly appreciate it.
I've tried using WaitForRing() in case it is bad signaling coming from
the phone company. But that has not helped.
I highly doubt this is the cause. I suspect a dialplan error.
Ask your client if they are doing anything else when this happens, like
making a fax call ;)
This setup has:
Asterisk 1.2.17
Zaptel (whatever was distributed same time as Asterisk 1.2.17)
CentOS 4.4
Polycom 301's throughout
Sangoma A200 with 2 ports connected to PSTN.
Thanks for any help.
We kinda need to see your dialplan to provide any useful help.
-Stephen-
Thanks for responding, Stephen.
The client has a fax line, but it is separate line from the Asterisk
box. They have 3 lines coming in. 1 goes directly to fax machine and 2
go to Asterisk box.
I've searched the archives and phantom ringing comes up a few times
without any real resolutions that I can see.
The only thing very different about this installation is that the
customer has no intermediate IVR. They want the phones to ring directly
to a group and if no answer, then go to a mini IVR that asks if they
would like to hold longer or leave a message.
The device calling on CallerID when this happens is the default
CallerID set in sip.conf Device callernum which was never changed.
That was changed to a correct value. It appears as though the system is
calling itself.
extensions.conf:
[incoming]
exten=s,1,WaitForRing(5)
exten=s,n,Answer()
exten=s,n,Ringing()
exten=s,n,SetMusicOnHold(default)
exten=s,n,Wait(1)
exten=s,n,Goto(check_time,s,1)
[check_time]
exten=s,1,Answer()
exten=s,2,GotoIfTime(08:30-17:00,mon-fri,*,*,?daytime,s,1)
exten=s,3,Goto(after_hours,s,1)
[daytime]
exten=s,1,Answer()
exten=s,2,Set(TIMEOUT(response)=1)
exten=s,3,Dial(${GRP_All},20,tr)
exten=s,4,Background(custom/no-answer)
exten=1,1,Set(loops=0)
exten=1,2,Wait(0.5)
exten=1,3,Goto(ring-all-with-moh,s,1)
exten=2,1,Voicemail(b116)
exten=2,2,Hangup()
exten=5,1,Voicemail(b116)
exten=5,2,Hangup()
exten=555,1,VoicemailMain()
exten=i,1,Background(invalid)
exten=i,2,Goto(ring-all-with-moh,s,1)
exten=t,1,Set(loops=0)
exten=t,2,Wait(0.5)
exten=t,3,Goto(ring-all-with-moh,s,1)
[ring-all-with-moh]
exten=s,1,Answer()
exten=s,2,Noop(Loops are: ${loops})
exten=s,3,Macro(DialExtenNoVM,116|60|tm)
exten=s,4,Set(loops=$[${loops}+1])
exten=s,5,GotoIf($[${loops} = 2 ]?timeout-from-loop,s,1)
exten=s,6,Background(custom/no-answer)
exten=1,1,Goto(ring-all-with-moh,s,1)
exten=2,1,Voicemail(b116)
exten=2,2,Hangup()
exten=i,1,Playback(invalid)
exten=i,2,Goto(ring-all-with-moh,s,1)
exten=t,1,Goto(ring-all-with-moh,s,1)
sip.conf:
[general]
allowexternalinvites=yes
allowguest=no
autocreatepeer=no
autodomain=no
bindaddr=0.0.0.0
callerid=device callernum
canreinvite=no
checkmwi=30
compactheaders=no
context=incoming
defaultexpirey=120
dtmfmode=rfc2833
dumphistory=no
externrefresh=30
ignoreregexpire=no
insecure=no
maxexpirey=3600
musicclass=default
nat=no
notifyringing=yes
pedantic=no
progressinband=never
promiscredir=no
qualify=no
recordhistory=no
registerattempts=30
registertimeout=30
relaxdtmf=no
rtautoclear=no
rtcachefriends=no
rtpholdtimeout=600
rtpkeepalive=0
rtptimeout=3600
rtupdate=yes
sendrpid=no
sipdebug=no
srvlookup=no
tos=none
trustrpid=no
useclientcode=no
usereqphone=no
callevents=no
disallow=all
allow=ulaw
[116]
context=super-user
type=friend
canreinvite=no
dtmfmode=rfc2833
callerid=Barbara 116
nat=no
port=5060
qualify=no
secret=xxx
host=dynamic
[EMAIL PROTECTED]
disallow=all
allow=ulaw
[117]
context=super-user
type=friend
canreinvite=no
dtmfmode=rfc2833
callerid=Bill 117
nat=no
port=5060
qualify=no
secret=xxx
host=dynamic
[EMAIL PROTECTED]
disallow=all
Re: [asterisk-users] Phantom Calls
Matt wrote: I too have seen what Rob is saying.. on a Sangoma card. It was an easy fix in the config, but I don't remember what it was.. but basically it was stray voltage. On 6/18/07, * Rob Schall* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: We were having phantom calls as well. In our case, we had 2 pots line running in our sangoma card, and when you dial out, would would wait for whomever to pickup. If you gave up waiting an hung the phone up (we also had 2 normal phones plugged into fxs ports), it wouldn't immediately receive the hangup signal. The call would connect, then asterisk would turn around and try to call us back. If the other side hungup because they just heard dead error, then when you'd repickup your call, it would also be dead air. Not sure if this is the same case as yours, but ours was odd as well. Rob Thanks for responding, Guys. I can't say if that is the behavior that causes it to happen, but I have asked the customer to take note of that. The symptoms that you both describe are exactly what they are experiencing so this is a welcome lead. Here is the zapata.conf below. [channels] usecallerid=yes cidsignalling=bell cidstart=ring usecallingpres=no echocancel=yes echocanclewhenbridged=yes echotraining=no rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 hanguponpolarityswitch=yes answeronpolarityswitch=no ringtimeout=8000 musiconhold=default busydetect=yes busycount=6 usecallerid=yes hidcallerid=no callwaiting=no threewaycalling=no transfer=no echocancel=yes echotraining=yes echocanelwhenbriged=yes context=incoming callprogress=no answeronpolarityswitch=no signalling=fxs_ks channel= 3 busydetect=yes busycount=6 usecallerid=yes hidcallerid=no callwaiting=no threewaycalling=no transfer=no echocancel=yes echotraining=yes echocanelwhenbriged=yes context=incoming callprogress=no answeronpolarityswitch=no signalling=fxs_ks channel= 4 -- Warm Regards, Lee ___ --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] Phantom calls: Detecting hangup quicker
On 07/06/07, Stephen Bosch [EMAIL PROTECTED] wrote: Gavin Henry wrote: Dear all, We seem to be getting phantom calls when a inbound caller via the legacy pbx hangups before the SIP handsets have answered. The extensions also seem to hear ringing on the lines too sometimes. SIP Inbound | legacy pbx (analogue) - (sangoma a400d) asterisk - SIP phones Basically if a user hangups before the call has bridged, I think. Is there anything we can do about this? Yet another call progress detection issue. Ah, sorry. I didn't know the right terms to search for beforehand. Analog lines are problematic this way. Search the archives for call progress detection or disconnect supervision. Many thanks. -Stephen- ___ --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
[asterisk-users] Phantom calls: Detecting hangup quicker
Dear all, We seem to be getting phantom calls when a inbound caller via the legacy pbx hangups before the SIP handsets have answered. The extensions also seem to hear ringing on the lines too sometimes. SIP Inbound | legacy pbx (analogue) - (sangoma a400d) asterisk - SIP phones Basically if a user hangups before the call has bridged, I think. Is there anything we can do about this? Thanks, Gavin. ___ --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] Phantom calls: Detecting hangup quicker
Gavin Henry wrote: Dear all, We seem to be getting phantom calls when a inbound caller via the legacy pbx hangups before the SIP handsets have answered. The extensions also seem to hear ringing on the lines too sometimes. SIP Inbound | legacy pbx (analogue) - (sangoma a400d) asterisk - SIP phones Basically if a user hangups before the call has bridged, I think. Is there anything we can do about this? Yet another call progress detection issue. Analog lines are problematic this way. Search the archives for call progress detection or disconnect supervision. -Stephen- ___ --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
