[asterisk-ss7] chan_ss7 incoming callbycall call problems

Simon Handschin simon.handschin at gmail.com
Fri Mar 2 23:58:17 CST 2012


Hi,

I tried it libss7 before, but there i've got problems receiving SAM messages.

2012/3/3 bipin singh <bipinraghuvanshi at gmail.com>:
> Hi ,
>             If possible try libss7.
>
>
> On Sat, Mar 3, 2012 at 2:06 AM, Simon Handschin <simon.handschin at gmail.com>
> wrote:
>>
>> Hi thank you for your fast answer.
>> Its about carrier pre selection eg i make a call from an external phone
>> (not connected to the server) with our cpc like 1234.
>>
>> 12340041441234567
>>
>> The Call will be routet to our server from the provider. They have enbloc
>> enabled.
>>
>> Dialplan is like exten => _12340041xxxxxxxxxx (for switzerland and i know
>> digits lenght)
>>
>> Its the # key because of the enbloc from our Provider?
>>
>> Greetings,
>>
>> Simon
>>
>> Am 02.03.2012 20:37 schrieb <florian at gruendler.net>:
>>
>>> Define "callbycall" number. What is the callflow?
>>> Accepted where, by what equipment?
>>> Are you trying to describe something which has to do with overlap vs.
>>> enbloc
>>> dialing in a call-through scenario with digit collection on Asterisk or
>>> is
>>> this a one stage call where the call enters and leaves Asterisk in one
>>> go?
>>> By what means is this ISDN phone connected to the Asterisk machine?
>>> How does the dialplan look like?
>>>
>>>
>>>
>>> > -----Ursprüngliche Nachricht-----
>>> > Von: asterisk-ss7-bounces at lists.digium.com [mailto:asterisk-ss7-
>>> > bounces at lists.digium.com] Im Auftrag von Simon Handschin
>>> > Gesendet: Freitag, 2. März 2012 20:07
>>> > An: asterisk-ss7 at lists.digium.com
>>> > Betreff: [asterisk-ss7] chan_ss7 incoming callbycall call problems
>>> >
>>> > 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
>>> >
>>> > --
>>> > _____________________________________________________________________
>>> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>> >
>>> > asterisk-ss7 mailing list
>>> > To UNSUBSCRIBE or update options visit:
>>> >    http://lists.digium.com/mailman/listinfo/asterisk-ss7
>>>
>>>
>>> --
>>> _____________________________________________________________________
>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>>
>>> asterisk-ss7 mailing list
>>> To UNSUBSCRIBE or update options visit:
>>>   http://lists.digium.com/mailman/listinfo/asterisk-ss7
>>
>>
>> --
>> _____________________________________________________________________
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> asterisk-ss7 mailing list
>> To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-ss7
>
>
>
>
> --
> BIPIN RAGHUVANSHI
> OPERATION HEAD
> ASTERISK (DEVELOPMENT AND RESEARCH)
> WWW.EHORIZONS.IN
> 011-32323262
> 011-46334633
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-ss7 mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-ss7



More information about the asterisk-ss7 mailing list