[Asterisk-Users] SIP extension calls itself intermittently

Rich Adamson radamson at routers.com
Fri Nov 4 22:22:22 MST 2005


> 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.





More information about the asterisk-users mailing list