[asterisk-ss7] chan_ss7 incoming callbycall call problems

Simon Handschin simon.handschin at gmail.com
Fri Mar 2 13:07:29 CST 2012


Hello everyone.

I have a problem. I have a callbycall number routet to our E1 Server.
If i Dial from my ISDN Phone to this callbycall number the call only
get acceptet if i press # key. Normal dial in and out works on our
server just if dial with call by call it did not work.

I Use

Sangoma A104 Card (Only 1 Port is Used)

Asterisk 1.8.9.3
Dahdi 2.6
chan_ss7 2.1.0
Wanpipe 3.5.25

dahdi systems.conf

#autogenerated by /usr/sbin/wancfg_dahdi do not hand edit
#autogenrated on 2012-03-02
#Dahdi Channels Configurations
#For detailed Dahdi options, view /etc/dahdi/system.conf.bak
loadzone=de
defaultzone=de

#Sangoma A104 port 1 [slot:3 bus:13 span:1] <wanpipe1>
span=1,0,0,ccs,hdb3,crc4
bchan=1-31
echocanceller=mg2,2-31
#hardhdlc=1

wanpipe1.conf


#================================================
# WANPIPE1 Configuration File
#================================================
#
# Date: Wed Dec  6 20:29:03 UTC 2006
#
# Note: This file was generated automatically
#       by /usr/local/sbin/setup-sangoma program.
#
#       If you want to edit this file, it is
#       recommended that you use wancfg program
#       to do so.
#================================================
# Sangoma Technologies Inc.
#================================================

[devices]
wanpipe1 = WAN_AFT_TE1, Comment

[interfaces]
w1g1 = wanpipe1, , TDM_VOICE, Comment

[wanpipe1]
CARD_TYPE 	= AFT
S514CPU 	= A
CommPort 	= PRI
AUTO_PCISLOT 	= NO
PCISLOT 	= 3
PCIBUS  	= 13
FE_MEDIA	= E1
FE_LCODE	= HDB3
FE_FRAME	= CRC4
FE_LINE		= 1
TE_CLOCK 	= MASTER
TE_REF_CLOCK    = 0
TE_SIG_MODE     = CCS
TE_HIGHIMPEDANCE	= NO
TE_RX_SLEVEL    = 430
HW_RJ45_PORT_MAP = DEFAULT
LBO 		= 120OH
FE_TXTRISTATE	= NO
MTU 		= 1500
UDPPORT		= 9000
TTL		= 255
IGNORE_FRONT_END	= NO
TDMV_SPAN		= 1
TDMV_DCHAN		= 0
TE_AIS_MAINTENANCE = NO         #NO: defualt  YES: Start port in AIS
Blue Alarm and keep line down
                                #wanpipemon -i w1g1 -c Ttx_ais_off to
disable AIS maintenance mode
								#wanpipemon -i w1g1 -c Ttx_ais_on to enable AIS maintenance mode
TDMV_HW_DTMF		= NO		# YES: receive dtmf events from hardware
TDMV_HW_FAX_DETECT		= NO		# YES: receive fax 1100hz events from hardware
HWEC_OPERATION_MODE     = OCT_NORMAL    # OCT_NORMAL: echo cancelation
enabled with nlp (default)
										# OCT_SPEECH: improves software tone detection by disabling
NLP (echo possible)
										# OCT_NO_ECHO:disables echo cancelation but allows VQE/tone
functions.
HWEC_DTMF_REMOVAL       = NO    # NO: default  YES: remove dtmf out of
incoming media (must have hwdtmf enabled)
HWEC_NOISE_REDUCTION    = NO    # NO: default  YES: reduces noise on
the line - could break fax
HWEC_ACUSTIC_ECHO       = NO    # NO: default  YES: enables acustic
echo cancelation
HWEC_NLP_DISABLE        = NO    # NO: default  YES: guarantees
software tone detection (possible echo)
HWEC_TX_AUTO_GAIN       = 0     # 0: disable   -40-0: default tx audio
level to be maintained (-20 default)
HWEC_RX_AUTO_GAIN       = 0     # 0: disable   -40-0: default tx audio
level to be maintained (-20 default)
HWEC_TX_GAIN            = 0		# 0: disable   -24-24: db values to be
applied to tx signal
HWEC_RX_GAIN            = 0		# 0: disable   -24-24: db values to be
applied to tx signal

[w1g1]
ACTIVE_CH	= ALL
TDMV_HWEC	= NO
MTU 		= 8


ss7.conf

[linkset-siuc]
enabled => yes
enable_st => no
use_connect => no
hunting_policy => even_mru
context => ss7
language => de
t35 => 4000,st
subservice => auto
variant => ITU


[link-l1]
linkset => siuc
channels => 2-31
schannel => 1
firstcic => 1
enabled => yes
sltm => no
sls=0
echocancel => no
echocan_train => 350
echocan_taps => 128

[host-SS7-1]
if-1 => 10.1.40.71
enabled => yes
opc => 1601
dpc => siuc:300
links => l1:1                         ;span 1 of dahdi/system.conf
;globaltitle => 0x00, 0x03, 0x01, 7773
ssn => 7


Anyone got an idea about this?

Greetings

Simon



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