Lists Pleasants
2005-Nov-04 18:59 UTC
[Asterisk-Users] SIP extension calls itself intermittently
Intermittently I'll get calls from my only SIP extension to itself via
the Zap/1. I have no clue and have found nothing online. I have listed
my configurations and a sample of the console messages I see why
debugging. Right now it only happens to the 6000 extension. Any
assistance is appreciated.
Thanks,
Chip
-- Starting simple switch on 'Zap/1-1'
Nov 4 14:00:54 WARNING[4156]: chan_zap.c:5476 ss_thread: CallerID
returned with error on channel 'Zap/1-1'
-- Executing Wait("Zap/1-1", "2") in new stack
-- Executing Answer("Zap/1-1", "") in new stack
-- Executing Dial("Zap/1-1", "SIP/6000|20") in new stack
-- Called 6000
-- SIP/6000-3d34 is ringing
== Spawn extension (from-pstn, s, 3) exited non-zero on 'Zap/1-1'
-- Executing Hangup("Zap/1-1", "") in new stack
== Spawn extension (from-pstn, h, 1) exited non-zero on 'Zap/1-1'
-- Hungup 'Zap/1-1'
bart*CLI>
extentions.conf
[general]
static=yes
writeprotect=no
[globals]
[default]
exten => _.,1,Dial(IAX2/chip:xxxxxx@172.16.96.247/${EXTEN})
exten => _.,2,Hangup
exten => _.,3,Congestion
exten => i,1,Hangup
exten => t,1,Hangup
exten => h,1,Hangup
exten => 5000,1,Answer
exten => 5000,2,Dial(SIP/4389,20)
exten => 5000,3,Voicemail(u4389)
exten => 5000,4,Hangup
exten => 9000,1,Answer
exten => 9000,2,VoiceMailMain(s${CALLERIDNUM})
exten => 9000,3,Hangup
[to-pstn]
exten => _.,1,Dial(Zap/1/${EXTEN})
exten => _.,2,Hangup
exten => _.,3,Congestion
exten => i,1,Hangup
exten => t,1,Hangup
exten => h,1,Hangup
[local]
exten => 6000,1,Answer
exten => 6000,2,Dial(SIP/6000,20)
exten => 6000,3,Hangup
exten => 6001,1,Answer
exten => 6001,2,Dial(SIP/6001,30)
exten => 6001,3,Hangup
exten => 9000,1,Answer
exten => 9000,2,VoiceMailMain(s${CALLERIDNUM})
exten => 9000,3,Hangup
exten => i,1,Hangup
exten => t,1,Hangup
exten => h,1,Hangup
[from-pstn]
exten => s,1,Wait(2)
exten => s,2,Answer
exten => s,3,Dial(SIP/6000,20)
exten => s,4,Voicemail(u6000)
exten => s,5,Congestion
exten => s,6,Hangup
exten => i,1,Hangup
exten => t,1,Hangup
exten => h,1,Hangup
[internal]
include => local
include => to-pstn
exten => i,1,Hangup
exten => t,1,Hangup
exten => h,1,Hangup
sip.conf
[general]
context=default
port=5060
bindaddr=0.0.0.0
srvlookup=yes
[4389]
type=friend
username=4389
secret=4389
host=dynamic
callerid="Chip xxxxx" <4389>
nat=no
disallow=all
allow=gsm
allow=ulaw
allow=alaw
mailbox=4389@default
[6000]
type=friend
username=6000
secret=6000
host=dynamic
callerid="xxxxx" <6000>
nat=no
disallow=all
allow=ulaw
allow=alaw
context=internal
mailbox=6000@internal
canreinvite=no
[6001]
type=friend
username=6001
secret=6001
host=dynamic
callerid="xxxxxxx" <6001>
nat=no
disallow=all
allow=gsm
allow=ulaw
allow=alaw
context=internal
mailbox=6001@internal
zaptel.conf
[channels]
language=en
context=from-pstn
switchtype=national
busydetect=yes
busycount=4
callprogress=yes
signalling=fxs_ks
rxwink=300
usecallerid=yes
cidsignalling=bell
cidstart=ring
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
echotraining=yes
echotraining=400
rxgain=0.0
txgain=0.0
group=1
callgroup=1
pickupgroup=1
immediate=no
callerid="name" <(xxx) xxx-xxxx>
busydetect=yes
busycount=4
callprogress=yes
channel => 1
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Rich Adamson
2005-Nov-04 22:22 UTC
[Asterisk-Users] SIP extension calls itself intermittently
> Intermittently Ill get calls from my only SIP extension to itself via the Zap/1. I have no clue and havefound nothing online. I have listed my configurations and a> sample of the console messages I see why debugging. Right now it only happens to the 6000 extension. Anyassistance is appreciated.>> [from-pstn] > exten => s,1,Wait(2) > exten => s,2,Answer > exten => s,3,Dial(SIP/6000,20) > exten => s,4,Voicemail(u6000) > exten => s,5,Congestion > exten => s,6,HangupIn the above, you don't want to "answer" and incoming call in your dialplan. When SIP/6000 picks up the phone, an answer is automatically sent back to the pstn. So, change the above to something like this: [from-pstn] exten => s,1,Dial(SIP/6000,20) exten => s,2,Voicemail(u6000) exten => s,102,Voicemail(b6000) exten => s,103,Hangup The following file is /etc/asterisk/zapata.conf, not zaptel.conf> zaptel.conf > [channels] > language=en > context=from-pstn > switchtype=national > busydetect=yes > busycount=4 > callprogress=yes > signalling=fxs_ks > rxwink=300 > usecallerid=yes > cidsignalling=bell > cidstart=ring > hidecallerid=no > callwaiting=yes > usecallingpres=yes > callwaitingcallerid=yes > threewaycalling=yes > transfer=yes > cancallforward=yes > callreturn=yes > echocancel=yes > echocancelwhenbridged=yes > echotraining=yes > echotraining=400 > rxgain=0.0 > txgain=0.0 > group=1 > callgroup=1 > pickupgroup=1 > immediate=no > callerid="name" <(xxx) xxx-xxxx> > busydetect=yes > busycount=4 > callprogress=yes > channel => 1Far too much junk and duplication in the above. Read the following including the comments. [channels] language=en busydetect=yes busycount=4 cidsignalling=bell cidstart=ring callerid=asreceived ; everything listed above applies to all channels defined below. Therefore ; only have to define them one time. ; the following channel definition is for the pstn line (channel 1) context=from-pstn ; switchtype=national ; this statement is for ISDN, not analog pstn. remove it ; busydetect=yes ; these two statements belong above and apply to all channels. ; busycount=4 callprogress=yes ; this should "probably" be =no signalling=fxs_ks ; rxwink=300 ; this statement isn't used with fxs_ks, remove it. usecallerid=yes ; cidsignalling=bell ; these two statements belong above and apply to all channels. ; cidstart=ring hidecallerid=no callwaiting=yes usecallingpres=yes ; not sure about the statement. remove it. callwaitingcallerid=yes threewaycalling=yes transfer=yes ; the pstn caller is _not_ going to transfer anything. remove it. cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=yes ; these two statements are identical. remove one of them. echotraining=400 rxgain=0.0 txgain=0.0 group=1 callgroup=1 ; this statement isn't needed, remove it. pickupgroup=1 ; the pstn line isn't going to pick up a call. remove it. immediate=no callerid="name" <(xxx) xxx-xxxx> ; not needed. remove it. callprogress=yes ; used this above. remove it. channel => 1 Cleaned up, your zapata.conf should look about like this: [channels] language=en busydetect=yes busycount=6 ; changed this from 4 to 6. cidsignalling=bell cidstart=ring callerid=asreceived ; the following channel definition is for the pstn line (channel 1) context=from-pstn callprogress=no ; changed this from yes to no. signalling=fxs_ks usecallerid=yes hidecallerid=no callwaiting=yes callwaitingcallerid=yes threewaycalling=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=400 rxgain=0.0 txgain=0.0 group=1 immediate=no channel => 1 Now, to answer your original question "after" cleaning up the statements, try busycount=6 and callprogress=no. What is likely happening is that either a bridged analog phone is connected to your pstn line and when someone is talking on that phone, asterisk interprets the audio as ringing. If there is no bridged analog phone, then * is likely interpreting noise of some sort as ringing. The callprogress=yes statement that you had doesn't really do anything for you, so either remove it or change it to callprogress=no. Several of the above statements that are left are simply repeating default values. If those statements were not there, it would function the same. But, while you're learning what these things do, I found it useful to leave them as is so as to remind you what those default values are actually in use.
David J Carter
2005-Nov-05 04:12 UTC
[Asterisk-Users] SIP extension calls itself intermittently
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Lists
Pleasants
Sent: 05 November 2005 01:59
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] SIP extension calls itself intermittently
Intermittently I?ll get calls from my only SIP extension to itself via the
Zap/1. I have no clue and have found nothing online. I have listed my
configurations and a sample of the console messages I see why debugging.
Right now it only happens to the 6000 extension.? Any assistance is
appreciated.
Thanks,
Chip
??? -- Starting simple switch on 'Zap/1-1'
Nov? 4 14:00:54 WARNING[4156]: chan_zap.c:5476 ss_thread: CallerID returned
with error on channel 'Zap/1-1'
??? -- Executing Wait("Zap/1-1", "2") in new stack
??? -- Executing Answer("Zap/1-1", "") in new stack
??? -- Executing Dial("Zap/1-1", "SIP/6000|20") in new stack
??? -- Called 6000
??? -- SIP/6000-3d34 is ringing
? == Spawn extension (from-pstn, s, 3) exited non-zero on 'Zap/1-1'
??? -- Executing Hangup("Zap/1-1", "") in new stack
? == Spawn extension (from-pstn, h, 1) exited non-zero on 'Zap/1-1'
??? -- Hungup 'Zap/1-1'
bart*CLI>
===========================================================================================================================================
Chip,
The output above looks to me as an incoming call.
I think you would have seen the SIP extension calling on ZAP/1-1 if
initiated from your end, and would not call in as the line was busy.
I get this now and again in the UK, usually in an evening time when the
Telco do an auto check of line status.
Regards
Dave
> Rich Adamson wrote: > > ; Calls directed to Teliax.com > > exten => _1NXXXXXXXXX,1,Set(CallerIDnum=4024325395|a) > > exten => _1NXXXXXXXXX,2,Set(CallerIDname=NPI|a) > > exten => _1NXXXXXXXXX,3,Dial(IAX2/teliaxout/${EXTEN}) > > This format is, of course, only on CVS-HEAD/1.2xOps, forgot to mention that. Thanks.
Lists Pleasants
2005-Nov-06 12:19 UTC
[Asterisk-Users] SIP extension calls itself intermittently
Thank you very much for the help! I continue to have the mystery calls
but not as often. I have attached the debugging info I captured. I also
removed every piece of equipment and have a single line coming from the
NIU into the X100P Clone card. I do have DSL so a DSL filter is required
along is in the middle. What actually causes Asterisk/ZAP to thick there
is a call?
Thanks,
Chip
Nov 6 08:31:12 VERBOSE[4851]: -- Starting simple switch on
'Zap/1-1'
Nov 6 08:31:20 WARNING[4851]: CallerID returned with error on channel
'Zap/1-1'
Nov 6 08:31:20 VERBOSE[4851]: -- Executing Dial("Zap/1-1",
"SIP/6000|20") in new stack
Nov 6 08:31:20 DEBUG[4851]: Setting NAT on RTP to 0
Nov 6 08:31:20 DEBUG[4851]: Outgoing Call for 6000
Nov 6 08:31:20 DEBUG[4851]: Call from user '6000' is 1 out of 0
Nov 6 08:31:20 VERBOSE[4851]: -- Called 6000
Nov 6 08:31:21 VERBOSE[4851]: -- SIP/6000-0821 is ringing
Nov 6 08:31:28 DEBUG[4851]: update_user_counter(6000) - decrement
outUse counter
Nov 6 08:31:28 DEBUG[4851]: Exiting with DIALSTATUS=CANCEL.
Nov 6 08:31:28 VERBOSE[4851]: == Spawn extension (from-pstn, s, 1)
exited non-zero on 'Zap/1-1'
Nov 6 08:31:28 VERBOSE[4851]: -- Executing Hangup("Zap/1-1",
"") in
new stack
Nov 6 08:31:28 VERBOSE[4851]: == Spawn extension (from-pstn, h, 1)
exited non-zero on 'Zap/1-1'
Nov 6 08:31:28 DEBUG[4851]: Hangup: channel: 1 index = 0, normal = 18,
callwait = -1, thirdcall = -1
Nov 6 08:31:28 DEBUG[4851]: disabled echo cancellation on channel 1
Nov 6 08:31:28 DEBUG[4851]: Set option TDD MODE, value: OFF(0) on
Zap/1-1
Nov 6 08:31:28 DEBUG[4851]: Updated conferencing on 1, with 0
conference users
Nov 6 08:31:28 VERBOSE[4851]: -- Hungup 'Zap/1-1'
Nov 6 08:31:28 DEBUG[4851]: Acked pending invite 102
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Rich
Adamson
Sent: Saturday, November 05, 2005 12:22 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] SIP extension calls itself intermittently
> Intermittently Ill get calls from my only SIP extension to itself via
the Zap/1. I have no clue and have
found nothing online. I have listed my configurations and
a> sample of the console messages I see why debugging. Right now it only
happens to the 6000 extension. Any
assistance is appreciated.>
> [from-pstn]
> exten => s,1,Wait(2)
> exten => s,2,Answer
> exten => s,3,Dial(SIP/6000,20)
> exten => s,4,Voicemail(u6000)
> exten => s,5,Congestion
> exten => s,6,Hangup
In the above, you don't want to "answer" and incoming call in your
dialplan. When SIP/6000 picks up the phone, an answer is automatically
sent back to the pstn. So, change the above to something like this:
[from-pstn]
exten => s,1,Dial(SIP/6000,20)
exten => s,2,Voicemail(u6000)
exten => s,102,Voicemail(b6000)
exten => s,103,Hangup
The following file is /etc/asterisk/zapata.conf, not
zaptel.conf> zaptel.conf
> [channels]
> language=en
> context=from-pstn
> switchtype=national
> busydetect=yes
> busycount=4
> callprogress=yes
> signalling=fxs_ks
> rxwink=300
> usecallerid=yes
> cidsignalling=bell
> cidstart=ring
> hidecallerid=no
> callwaiting=yes
> usecallingpres=yes
> callwaitingcallerid=yes
> threewaycalling=yes
> transfer=yes
> cancallforward=yes
> callreturn=yes
> echocancel=yes
> echocancelwhenbridged=yes
> echotraining=yes
> echotraining=400
> rxgain=0.0
> txgain=0.0
> group=1
> callgroup=1
> pickupgroup=1
> immediate=no
> callerid="name" <(xxx) xxx-xxxx>
> busydetect=yes
> busycount=4
> callprogress=yes
> channel => 1
Far too much junk and duplication in the above. Read the following
including the comments.
[channels]
language=en
busydetect=yes
busycount=4
cidsignalling=bell
cidstart=ring
callerid=asreceived
; everything listed above applies to all channels defined below.
Therefore
; only have to define them one time.
; the following channel definition is for the pstn line (channel 1)
context=from-pstn
; switchtype=national ; this statement is for ISDN, not analog pstn.
remove it
; busydetect=yes ; these two statements belong above and apply to all
channels.
; busycount=4
callprogress=yes ; this should "probably" be =no
signalling=fxs_ks
; rxwink=300 ; this statement isn't used with fxs_ks, remove it.
usecallerid=yes
; cidsignalling=bell ; these two statements belong above and apply to
all channels.
; cidstart=ring
hidecallerid=no
callwaiting=yes
usecallingpres=yes ; not sure about the statement. remove it.
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes ; the pstn caller is _not_ going to transfer anything.
remove it.
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
echotraining=yes ; these two statements are identical. remove one of
them.
echotraining=400
rxgain=0.0
txgain=0.0
group=1
callgroup=1 ; this statement isn't needed, remove it.
pickupgroup=1 ; the pstn line isn't going to pick up a call. remove it.
immediate=no
callerid="name" <(xxx) xxx-xxxx> ; not needed. remove it.
callprogress=yes ; used this above. remove it.
channel => 1
Cleaned up, your zapata.conf should look about like this:
[channels]
language=en
busydetect=yes
busycount=6 ; changed this from 4 to 6.
cidsignalling=bell
cidstart=ring
callerid=asreceived
; the following channel definition is for the pstn line (channel 1)
context=from-pstn
callprogress=no ; changed this from yes to no.
signalling=fxs_ks
usecallerid=yes
hidecallerid=no
callwaiting=yes
callwaitingcallerid=yes
threewaycalling=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
echotraining=400
rxgain=0.0
txgain=0.0
group=1
immediate=no
channel => 1
Now, to answer your original question "after" cleaning up the
statements, try busycount=6 and callprogress=no.
What is likely happening is that either a bridged analog phone is
connected to your pstn line and when someone is talking on that
phone, asterisk interprets the audio as ringing. If there is no
bridged analog phone, then * is likely interpreting noise of some
sort as ringing.
The callprogress=yes statement that you had doesn't really do anything
for you, so either remove it or change it to callprogress=no.
Several of the above statements that are left are simply repeating
default values. If those statements were not there, it would function
the same. But, while you're learning what these things do, I found it
useful to leave them as is so as to remind you what those default
values are actually in use.
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