Crossing calls

[XyVy]

Hi guys...

I am using elastix 1.1-7, asterisk 1.4.19 and misdn 1.1.7.2

When I do a call from extension X to outside and begin to talk, if other extension (for example Y), try to call to outside, in a few seconds the extension Y heards the first call from the extesion X, and X lost your call.

What can I do ?

Thanks.

The misdn.log shows:

Mon Jul 21 23:28:19 2008: P[ 1]  read_config: Getting Config
Mon Jul 21 23:28:19 2008: P[ 1]  * CALL: g:out1/968937299
Mon Jul 21 23:28:19 2008: P[ 1]  I SEND:SETUP oad:968898420 dad:968937299 pid:5
Mon Jul 21 23:28:19 2008: P[ 1]  I IND :NEW_CHANNEL oad:968898420 dad:968937299 pid:5 state:CALLING
Mon Jul 21 23:28:19 2008: P[ 1]  I IND :SETUP_ACKNOWLEDGE oad:968898420 dad:968937299 pid:5 state:CALLING
Mon Jul 21 23:28:21 2008: P[ 1]  I IND :CONNECT oad:968898420 dad:968937299 pid:5 state:CALLING_ACKNOWLEDGE
Mon Jul 21 23:28:21 2008: P[ 1]  I SEND:CONNECT_ACKNOWLEDGE  oad:968898420 dad:968937299 pid:5
Mon Jul 21 23:28:21 2008: P[ 1]   --> * Unknown Indication:20 pid:5
Mon Jul 21 23:28:44 2008: P[ 1]  read_config: Getting Config
Mon Jul 21 23:28:44 2008: P[ 1]  * CALL: g:out1/655052415
Mon Jul 21 23:28:44 2008: P[ 1]  I SEND:SETUP oad:968898420 dad:655052415 pid:6
Mon Jul 21 23:28:44 2008: P[ 1]  I IND :NEW_CHANNEL oad:968898420 dad:655052415 pid:6 state:CALLING
Mon Jul 21 23:28:44 2008: P[ 1]  I IND :SETUP_ACKNOWLEDGE oad:968898420 dad:655052415 pid:6 state:CALLING
Mon Jul 21 23:28:51 2008: P[ 1]  * IND : HANGUP pid:6 ctx:from-pstn dad:655052415 oad:0655052415 State:CALLING_ACKNOWLEDGE
Mon Jul 21 23:28:51 2008: P[ 1]  I SEND:DISCONNECT oad:968898420 dad:655052415 pid:6
Mon Jul 21 23:28:51 2008: P[ 1]  I IND :RELEASE oad:968898420 dad:655052415 pid:6 state:CLEANING
Mon Jul 21 23:28:51 2008: P[ 1]  Cannot hangup chan, no ast
Mon Jul 21 23:28:51 2008: P[ 1]  I IND :DISCONNECT oad:968898420 dad:968937299 pid:5 state:CONNECTED
Mon Jul 21 23:28:51 2008: P[ 1]  I SEND:RELEASE oad:968898420 dad:968937299 pid:5
Mon Jul 21 23:28:51 2008: P[ 1]  * IND : HANGUP pid:5 ctx:from-pstn dad: oad:0968937299 State:CONNECTED
Mon Jul 21 23:28:51 2008: P[ 1]  I IND :RELEASE_COMPLETE oad:968898420 dad:968937299 pid:5 state:CLEANING
Mon Jul 21 23:28:51 2008: P[ 1]  Cannot hangup chan, no ast

My misdn.conf:

<mISDNconf>
        <module poll="128" debug="0" timer="no">hfcmulti</module>
        <module debug="0" options="0">mISDN_dsp</module>
        <devnode user="asterisk" group="asterisk" mode="660">mISDN</devnode>
        <card type="BN4S0" master-clock="yes">
                <port mode="te" link="ptp">1</port>
                <port mode="te" link="ptp">2</port>
                <port mode="te" link="ptmp">3</port>
                <port mode="te" link="ptmp">4</port>
        </card>
</mISDNconf>

My /etc/asterisk/misdn.conf

; here you can define which bearers should be allowed
;
allowed_bearers=all

; Prefixes for national and international, those are put before the
; oad if an according dialplan is set by the other end.
;
; default values: nationalprefix      : 0
;                 internationalprefix : 00
;
nationalprefix=0
internationalprefix=00

; set rx/tx gains between -8 and 8 to change the RX/TX Gain
;
; default values: rxgain: 0
;                 txgain: 0
;
rxgain=0
txgain=0

; some telcos especially in NL seem to need this set to yes, also in
; switzerland this seems to be important
;
; default value: no
;
te_choose_channel=no

; This option defines, if chan_misdn should check the L1 on  a PMP
; before making a group call on it. The L1 may go down for PMP Ports
; so we might need this.
; But be aware! a broken or plugged off cable might be used for a group call
; as well, since chan_misdn has no chance to distinguish if the L1 is down
; because of a lost Link or because the Provider shut it down...
;
; default: no
;
pmp_l1_check=no


;
; in PMP this option defines which cause should be sent out to
; the 3. caller. chan_misdn does not support callwaiting on TE
; PMP side. This allows to modify the RELEASE_COMPLETE cause
; at least.
;
reject_cause=16


;
; Send Setup_Acknowledge on incoming calls anyway (instead of PROCEEDING),
; this requests additional Infos, so we can waitfordigits
; without much issues. This works only for PTP Ports
;
; default value: no
;
need_more_infos=no


;
; set this to yes if you want to disconnect calls when a timeout occurs
; for example during the overlapdial phase
;
nttimeout=no

; set the method to use for channel selection:
;   standard    - always choose the first free channel with the lowest number
;   round_robin - use the round robin algorithm to select a channel. use this
;                 if you want to balance your load.
;
; default value: standard
;
method=standard


; specify if chan_misdn should collect digits before going into the
; dialplan, you can choose yes=4 Seconds, no, or specify the amount
; of seconds you need;
;
overlapdial=yes

;
; dialplan means Type Of Number in ISDN Terms (for outgoing calls)
;
; there are different types of the dialplan:
;
; dialplan -> outgoing Number
; localdialplan -> callerid
; cpndialplan -> connected party number
;
; dialplan options:
;
; 0 - unknown
; 1 - International
; 2 - National
; 4 - Subscriber
;
; This setting is used for outgoing calls
;
; default value: 0
;
dialplan=0
localdialplan=0
cpndialplan=0

;
; turn this to no if you don't mind correct handling of Progress Indicators
;
early_bconnect=no


;
; turn this on if you like to send Tone Indications to a Incoming
; isdn channel on a TE Port. Rarely used, only if the Telco allows
; you to send indications by yourself, normally the Telco sends the
; indications to the remote party.
;
; default: no
;
incoming_early_audio=no

; uncomment the following to get into s extension at extension conf
; there you can use DigitTimeout if you can't or don't want to use
; isdn overlap dial.
; note: This will jump into the s exten for every exten!
;
; default value: no
;
;always_immediate=no

;
; set this to yes if you want to generate your own dialtone
; with always_immediate=yes, else chan_misdn generates the dialtone
;
; default value: no
;
nodialtone=no


; uncomment the following if you want callers which called exactly the
; base number (so no extension is set) jump to the s extension.
; if the user dials something more it jumps to the correct extension
; instead
;
; default value: no
;
;immediate=no

; uncomment the following to have hold and retrieve support
;
; default value: no
;
;hold_allowed=yes

; Pickup and Callgroup
;
; default values: not set = 0
; range: 0-63
;
callgroup=1
pickupgroup=1


;
; these are the exact isdn screening and presentation indicators
; if -1 is given for both values the presentation indicators are used
; from asterisks SetCallerPres application.
; s=0, p=0 -> callerid presented not screened
; s=1, p=1 -> callerid presented but screened (the remote end does not see it!)
;
; default values s=-1, p=-1
; estaba en -1 y -1
presentation=1
screen=1

; this enables echocancellation, with the given number of taps
; be aware, move this setting only to outgoing portgroups!
; A value of zero turns echocancellation off.
;
; possible values are: 0,32,64,128,256,yes(=128),no(=0)
;
; default value: no
;
; cambio a no
echocancel=yes

; Set this to no to disable echotraining. You can enter a number > 10
; the value is a multiple of 0.125 ms.
;
; default value: no
; yes = 2000
; no = 0
;
echotraining=no

;
; chan_misdns jitterbuffer, default 4000
;
jitterbuffer=4000

;
; change this threshold to enable dejitter functionality
;
jitterbuffer_upper_threshold=0


;
; change this to yes, if you want to bridge a mISDN data channel to
; another channel type or to an application.
;
hdlc=no


;
; defines the maximum amount of incoming calls per port for
; this group. Calls which exceed the maximum will be marked with
; the channel variable MAX_OVERFLOW. It will contain the amount of
; overflowed calls
;
max_incoming=2

;
; defines the maximum amount of outgoing calls per port for this group
; exceeding calls will be rejected
;
max_outgoing=2

;-------------------------------
; quito esto
[intern]
; define your ports, e.g. 1,2 (depends on mISDN-driver loading order)
;ports=1,2
; context where to go to when incoming Call on one of the above ports
;context=Intern

;[internPP]
;
; adding the postfix 'ptp' to a port number is obsolete now, chan_misdn
; parses /etc/misdn-init.conf and sets the ptp mode to the corresponding
; configs. For backwards compatibility you can still set ptp here.
;
;ports=3

;[first_extern]
; again port defs
;ports=4
; again a context for incoming calls
;context=Extern1
; msns for te ports, listen on those numbers on the above ports, and
; indicate the incoming calls to asterisk
; here you can give a comma separated list or simply an '*' for
; any msn.
;msns=*

; here an example with given msns
;[second_extern]
;ports=5
;context=Extern2
;callerid=15
;msns=102,144,101,10

; hasta aqui
;---------------------------------

[out1]
ports=1ptp
context=from-pstn
msns=*

[out2]
ports=2ptp
context=from-pstn

I can do calls and receive calls, but with the above problem.

I need a miracle!! (Excuse me but I am gonna crazy)!

Thanks from Spain!

 

[XyVy]

I see many misdn.conf and all has max_incoming and max_outgoing in -1 value, I wrote the value 2 thinking in 2 channels by port, but all configurations that I see in internet, they are in -1 value.

Could be the problem? What means -1 value?

Thanks in advance!

 

[foschi]

Hi,

thanks for your participation. Please do use the

-tag to include code for better readability.

 

[XyVy]

Okey, excuse me...

Any sugestion to my problem ?

thanks.

 

[XyVy]

The provider told me that configuration was wrong in one of the BRI access.

Now all is good!

Thanks!

 

[Burmann]

I think (but not tested) the max_incoming/max_outgoing value of -1 is for "no limit".
I don't know if the call is automaticaly rejected or only set the variable MAX_OVERFLOW.
You can test it and post the result here .