[asterisk-ss7] chan_ss7 incoming callbycall call problems
florian at gruendler.net
florian at gruendler.net
Sat Mar 3 13:24:24 CST 2012
Hi Simon
Yes, the en-bloc behaviour is then clearly induced by your inbound transit
carrier. Swisscom will do overlap by default after a unique routing
direction is evident (1XXX0XX) whereas they have different filter sets to
route value added and emergency numbers to an NPRN or not so they need to
collect the first 7 digits before the routing decision is made. But if your
equipment is unable to handle overlap routing table lookups and route
selection, your transit provider is probably doing digit collection and
either waits for a # to send you whatever is in the buffer or has an defined
interdigit timeout of n seconds. Put an analyzer on your trunk and capture
the SS7 MTP3 to see whether SAMs are being transmitted or you see an f at
the end of the address indicating forced dialing or a complete enbloc
address. If you have a DMS, it has a feature to define digit count per
prefix, which speeds up the post dial delay for destinations with
standardized number lengths, such as Switzerland or the US. Unfortunately,
there is countries with a numbering plan having variable number sizes, the
worst I know being Austria, having numbers as short as +43XXXXX given to
customers and some with up to 20+ digits. It is a clear quality
characteristic to have overlap dialing end to end with the incumbent carrier
to send an ACM when the number is found in a local PBX. However, even some
major labels are unable to handle overlap and totally cripple the quality of
service giving high PDD. Only selected equipment supports overlap over SIP
because of interworking issues.
I even don’t know whether Asterisk supports that today. I have found a
recent discussion about RFC4497:
http://lists.digium.com/pipermail/asterisk-users/2011-September/266009.html
Cisco does:
http://www.cisco.com/en/US/docs/voice_ip_comm/pgw/9/feature/module/9.7_3_/si
p_overlap.html#wp1054594
Regards, Florian
(consulting Swiss OLOs as well)
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 21:37
An: asterisk-ss7 at lists.digium.com
Betreff: Re: [asterisk-ss7] chan_ss7 incoming callbycall call problems
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
>
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