[asterisk-ss7] Voice is not coming in Outbound Isup Call
Rajesh Mahajan
rajeshmahajan09 at gmail.com
Fri Sep 18 05:01:35 CDT 2009
How to solve this problem ?
On Fri, Sep 18, 2009 at 1:16 PM, <asterisk-ss7-request at lists.digium.com> wrote:
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> Today's Topics:
>
> 1. Re: handling * and # of dialed number on the extension.conf
> (Rafael Visser)
> 2. SS7 for Verisign A-Link, M3UA? (James Wiegand)
> 3. Voice is not coming in Outbound Isup Call (Rajesh Mahajan)
> 4. Re: Voice is not coming in Outbound Isup Call (Wasim Baig)
> 5. Re: handling * and # of dialed number on the extension.conf
> (Kaloyan Kovachev)
> 6. Re: Voice is not coming in Outbound Isup Call (Attila Domjan)
>
>
> ----------------------------------------------------------------------
>
> Message: 1
> Date: Thu, 17 Sep 2009 14:32:33 -0400
> From: Rafael Visser <visser.rafael at gmail.com>
> Subject: Re: [asterisk-ss7] handling * and # of dialed number on the
> extension.conf
> To: asterisk-ss7 at lists.digium.com
> Message-ID:
> <b1b91df00909171132q6d20a908if4b012c703f5c788 at mail.gmail.com>
> Content-Type: text/plain; charset=ISO-8859-1
>
> Gustavo:
> Are you talking about chan_ss7 or libss7?
> I think that it would help on chan_ss7.
>
> I am not getting the same results with libss7.
> Or perhaps i'm doing wrong in other place..
>
>
>
>
>
> 2009/9/17, Gustavo Marsico <gustavomarsico at gmail.com>:
>> * is B, and # is C.
>> Replace them and it should be fine.
>>
>> Regards,
>>
>> Gustavo
>>
>>
>> On 17 Sep 2009, at 09:43, Rafael Visser wrote:
>>
>>> Hi guys.
>>>
>>> I use asterisk with libss7 as an ivr for vas purpose on a mobile
>>> company.
>>>
>>> Some of the numbers to access the service begins with * or # like
>>> "*555".
>>>
>>> When we access the services from a sip home, the "*" are interpreted
>>> in the dial plan fine.
>>> But when we access from mobile phone through libss7, asterisk can't
>>> interprete the dialed number.
>>>
>>> Is there some trick to handle "*" or "#" on the dni with libss7 and
>>> asterisk?.
>>>
>>> thanks in advance!!!
>>>
>>>
>>>
>>> this is the the debug of one call.
>>> Len = 73 [ 97 96 3f 85 e5 09 71 f2 5f 00 01 00 60 01 0a 00 02 08 06 83
>>> 90 3b 38 87 0f 0a 07 02 13 90 17 02 10 86 08 01 00 1d 03 80 90 a3 31
>>> 02 00 64 3a 06 44 05 95 00 00 00 3f 08 04 93 95 95 17 02 00 87 39 06
>>> 31 d0 3a d0 3f c0 00 ]
>>> FSN: 22 FIB 1
>>> BSN: 23 BIB 1
>>> <[1] MSU
>>> [ 97 96 3f ]
>>> Network Indicator: 2 Priority: 0 User Part: ISUP (5)
>>> [ 85 ]
>>> OPC XXXX DPC XXXX SLS 15
>>> [ e5 09 71 f2 ]
>>> CIC: 95
>>> [ 5f 00 ]
>>> Message Type: IAM
>>> [ 01 ]
>>> --FIXED LENGTH PARMS[4]--
>>> Nature of Connection Indicator:
>>> Satellites in connection: 0
>>> Continuity Check: Check not required (0)
>>> Outgoing half echo control device: not
>>> included (0)
>>> [ 00 ]
>>> Forward Call Indicators:
>>> Nat/Intl Call Ind: call to be treated as a
>>> national call (0)
>>> End to End Method Ind: no end-to-end method(s)
>>> available (0)
>>> Interworking Ind: no
>>> interworking encountered (0)
>>> End to End Info Ind: no end-to-end information
>>> available (0)
>>> ISDN User Part Ind: ISDN user part used all
>>> the way (1)
>>> ISDN User Part Pref Ind: ISDN
>>> user part not preferred all the way (1)
>>> ISDN Access Ind: originating access ISDN (1)
>>> SCCP Method Ind: no indication (0)
>>> [ 60 01 ]
>>> Calling Party's Category:
>>> Category: Ordinary calling subscriber (10)
>>> [ 0a ]
>>> Transmission Medium Requirements:
>>> Speech (0)
>>> [ 00 ]
>>> --VARIABLE LENGTH PARMS[1]--
>>> Called Party Number:
>>> Nature of address: 3
>>> NI: 1
>>> Numbering plan: 1
>>> Address signals:
>>> [ 06 83 90 3b 38 87 0f ]
>>> --OPTIONAL PARMS--
>>> Calling Party Number:
>>> Nature of address: 2
>>> NI: 0
>>> Numbering plan: 1
>>> Presentation: 0
>>> Screening: 3
>>> Address signals: 0971200199
>>> [ 0a 07 02 13 90 17 02 10 86 ]
>>> Optional forward call indicator:
>>> [ 08 01 00 ]
>>> User Service Information:
>>> [ 1d 03 80 90 a3 ]
>>> Propagation Delay Counter:
>>> Delay: 0ms
>>> [ 31 02 00 64 ]
>>> Unknown Parameter (0x3a):
>>> [ 44 05 95 00 00 00 ]
>>> Location Number:
>>> [ 3f 08 04 93 95 95 17 02 00 87 ]
>>> Parameter Compatibility Information:
>>> [ 39 06 31 d0 3a d0 3f c0 ]
>>>
>>> Unhandled optional parameter 0x8 'Optional forward call indicator'
>>> [0x0 ]
>>> Unhandled optional parameter 0x31 'Propagation Delay Counter'
>>> [0x0 0x64 ]
>>> Unhandled optional parameter 0x3a 'Unknown'
>>> [0x44 0x5 0x95 0x0 0x0 0x0 ]
>>> Unhandled optional parameter 0x3f 'Location Number'
>>> 0x4 0x93 0x95 0x95 0x17 0x2 0x0 0x87 ]
>>> Unhandled optional parameter 0x39 'Parameter Compatibility
>>> Information'
>>> [0x31 0xd0 0x3a 0xd0 0x3f 0xc0 ]
>>> Len = 16 [ 96 98 0d 85 c4 49 79 f2 5f 00 0c 02 00 02 81 81 ]
>>> FSN: 24 FIB 1
>>> BSN: 22 BIB 1
>>>> [1] MSU
>>> [ 96 98 0d
>>>
>>> _______________________________________________
>>> --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
>>
>
>
>
> ------------------------------
>
> Message: 2
> Date: Thu, 17 Sep 2009 17:42:49 -0500
> From: James Wiegand <originaljimdandy at gmail.com>
> Subject: [asterisk-ss7] SS7 for Verisign A-Link, M3UA?
> To: asterisk-ss7 at lists.digium.com
> Message-ID:
> <cb0ab51a0909171542j24e6fba1j8bf6f5c399b380e6 at mail.gmail.com>
> Content-Type: text/plain; charset=ISO-8859-1
>
> Hi,
>
> I'm new to all this SS7 stuff and we need to get Verisign working on
> Asterisk. What is the general cookbook for getting this going,
> assuming Asterisk/SS7/M3UA is a workable option?
>
> Thanks in advance,
> -jim
>
> --
> --
> Jim Wiegand
> -----------
> Home: originaljimdandy at gmail.com
> AIM: originaljimdandy
>
>
>
> ------------------------------
>
> Message: 3
> Date: Fri, 18 Sep 2009 11:41:52 +0530
> From: Rajesh Mahajan <rajeshmahajan09 at gmail.com>
> Subject: [asterisk-ss7] Voice is not coming in Outbound Isup Call
> To: asterisk-ss7 at lists.digium.com
> Message-ID:
> <c9961d450909172311o3c36da4wcd51b0580242d9a6 at mail.gmail.com>
> Content-Type: text/plain; charset=ISO-8859-1
>
> Hi All.
>
> We are using Sangoma A104u Quad Card for SS7.
>
> Incoming call is working fine.
> While in outbound call is working fine but not able to hear voice on
> the channel.
>
> Below is the config files
>
> chan_dahdi.conf
>
> [channels]
> ;switchtype=euroisdn
> usecallerid=yes
> callwaiting=yes
> usecallingpres=yes
> callwaitingcallerid=yes
> threewaycalling=yes
> transfer=yes
> canpark=yes
> cancallforward=yes
> callreturn=yes
> echocancel=yes
> echocancelwhenbridged=yes
> group=1
> callgroup=1
> pickupgroup=1
>
>
> signalling = ss7
> ss7type = itu
> ss7_called_nai=dynamic
> ss7_calling_nai=dynamic
> networkindicator=national
>
> ; port 1
> linkset = 1
> group = 1
> signalling=ss7
> ss7type = itu
> context = dialout
> pointcode = 8002
> adjpointcode = 9146
> defaultdpc = 9146
> networkindicator = national
> sigchan = 16
> cicbeginswith = 1
> channel => 1-15
> cicbeginswith = 17
> channel => 17-31
>
>
> /etc/dahdi/system.conf
>
> loadzone=us
> defaultzone=us
>
> #Sangoma A104 port 1 [slot:1 bus:12 span:1] <wanpipe1>
> span=1,0,0,ccs,hdb3
> bchan=1-15,17-31
> echocanceller=mg2,1-15,17-31
> #hardhdlc=16
> dchan=16
>
> /etc/wanpipe/wanpipe1.conf
> [devices]
> wanpipe1 = WAN_AFT_TE1, Comment
>
> [interfaces]
> w1g1 = wanpipe1, , TDM_VOICE, Comment
>
> [wanpipe1]
> CARD_TYPE = AFT
> S514CPU = A
> CommPort = PRI
> AUTO_PCISLOT = NO
> PCISLOT = 1
> PCIBUS = 12
> FE_MEDIA = E1
> FE_LCODE = HDB3
> FE_FRAME = NCRC4
> FE_LINE = 1
> TE_CLOCK = NORMAL
> TE_REF_CLOCK = 0
> TE_SIG_MODE = CCS
> TE_HIGHIMPEDANCE = NO
> LBO = 120OH
> FE_TXTRISTATE = NO
> MTU = 1500
> UDPPORT = 9000
> TTL = 255
> IGNORE_FRONT_END = NO
> TDMV_SPAN = 1
> TDMV_DCHAN = 0
> TDMV_HW_DTMF = NO
> TDMV_HW_FAX_DETECT = NO
>
> [w1g1]
> ACTIVE_CH = ALL
> TDMV_HWEC = NO
>
>
>
> ------------------------------
>
> Message: 4
> Date: Fri, 18 Sep 2009 12:19:31 +0600
> From: Wasim Baig <wasim at convergence.pk>
> Subject: Re: [asterisk-ss7] Voice is not coming in Outbound Isup Call
> To: asterisk-ss7 at lists.digium.com
> Message-ID:
> <b8ad2a5b0909172319j2de6f2e1p9eb75b2fca5b6c1d at mail.gmail.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> rajesh:
>
> use dahdi_monitor to see if the voice is actually going out on the
> particular channel
> or one above or below it, as its probably just a cic mismatch
>
> -wasim
>
> On Fri, Sep 18, 2009 at 12:11 PM, Rajesh Mahajan
> <rajeshmahajan09 at gmail.com>wrote:
>
>> Hi All.
>>
>> We are using Sangoma A104u Quad Card for SS7.
>>
>> Incoming call is working fine.
>> While in outbound call is working fine but not able to hear voice on
>> the channel.
>>
>> Below is the config files
>>
>> chan_dahdi.conf
>>
>> [channels]
>> ;switchtype=euroisdn
>> usecallerid=yes
>> callwaiting=yes
>> usecallingpres=yes
>> callwaitingcallerid=yes
>> threewaycalling=yes
>> transfer=yes
>> canpark=yes
>> cancallforward=yes
>> callreturn=yes
>> echocancel=yes
>> echocancelwhenbridged=yes
>> group=1
>> callgroup=1
>> pickupgroup=1
>>
>>
>> signalling = ss7
>> ss7type = itu
>> ss7_called_nai=dynamic
>> ss7_calling_nai=dynamic
>> networkindicator=national
>>
>> ; port 1
>> linkset = 1
>> group = 1
>> signalling=ss7
>> ss7type = itu
>> context = dialout
>> pointcode = 8002
>> adjpointcode = 9146
>> defaultdpc = 9146
>> networkindicator = national
>> sigchan = 16
>> cicbeginswith = 1
>> channel => 1-15
>> cicbeginswith = 17
>> channel => 17-31
>>
>>
>> /etc/dahdi/system.conf
>>
>> loadzone=us
>> defaultzone=us
>>
>> #Sangoma A104 port 1 [slot:1 bus:12 span:1] <wanpipe1>
>> span=1,0,0,ccs,hdb3
>> bchan=1-15,17-31
>> echocanceller=mg2,1-15,17-31
>> #hardhdlc=16
>> dchan=16
>>
>> /etc/wanpipe/wanpipe1.conf
>> [devices]
>> wanpipe1 = WAN_AFT_TE1, Comment
>>
>> [interfaces]
>> w1g1 = wanpipe1, , TDM_VOICE, Comment
>>
>> [wanpipe1]
>> CARD_TYPE = AFT
>> S514CPU = A
>> CommPort = PRI
>> AUTO_PCISLOT = NO
>> PCISLOT = 1
>> PCIBUS = 12
>> FE_MEDIA = E1
>> FE_LCODE = HDB3
>> FE_FRAME = NCRC4
>> FE_LINE = 1
>> TE_CLOCK = NORMAL
>> TE_REF_CLOCK = 0
>> TE_SIG_MODE = CCS
>> TE_HIGHIMPEDANCE = NO
>> LBO = 120OH
>> FE_TXTRISTATE = NO
>> MTU = 1500
>> UDPPORT = 9000
>> TTL = 255
>> IGNORE_FRONT_END = NO
>> TDMV_SPAN = 1
>> TDMV_DCHAN = 0
>> TDMV_HW_DTMF = NO
>> TDMV_HW_FAX_DETECT = NO
>>
>> [w1g1]
>> ACTIVE_CH = ALL
>> TDMV_HWEC = NO
>>
>> _______________________________________________
>> --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
>>
>
>
>
> --
> wasim h. baig | principal consultant | convergence pk | +92 300 8508070 |
> peace be upon you ...
> Sent from Lahore, Pakistan
> -------------- next part --------------
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> ------------------------------
>
> Message: 5
> Date: Fri, 18 Sep 2009 09:48:19 +0300
> From: "Kaloyan Kovachev" <kkovachev at varna.net>
> Subject: Re: [asterisk-ss7] handling * and # of dialed number on the
> extension.conf
> To: asterisk-ss7 at lists.digium.com
> Message-ID: <20090918064231.M36591 at varna.net>
> Content-Type: text/plain; charset=windows-1251
>
> Hi,
> for libss7 there two functions in isup.c that are responsible for this and
> they do not have ABCD*
> Look for char2digit and digit2char in isup.c and add the codes you need.
> Looking at the "Called Party Number: ... Address signals:" in your debug you
> should probably add "case 11: return '*'" in digit2char
>
> On Thu, 17 Sep 2009 14:32:33 -0400, Rafael Visser wrote
>> Gustavo:
>> Are you talking about chan_ss7 or libss7?
>> I think that it would help on chan_ss7.
>>
>> I am not getting the same results with libss7.
>> Or perhaps i'm doing wrong in other place..
>>
>> 2009/9/17, Gustavo Marsico <gustavomarsico at gmail.com>:
>> > * is B, and # is C.
>> > Replace them and it should be fine.
>> >
>> > Regards,
>> >
>> > Gustavo
>> >
>> >
>> > On 17 Sep 2009, at 09:43, Rafael Visser wrote:
>> >
>> >> Hi guys.
>> >>
>> >> I use asterisk with libss7 as an ivr for vas purpose on a mobile
>> >> company.
>> >>
>> >> Some of the numbers to access the service begins with * or # like
>> >> "*555".
>> >>
>> >> When we access the services from a sip home, the "*" are interpreted
>> >> in the dial plan fine.
>> >> But when we access from mobile phone through libss7, asterisk can't
>> >> interprete the dialed number.
>> >>
>> >> Is there some trick to handle "*" or "#" on the dni with libss7 and
>> >> asterisk?.
>> >>
>> >> thanks in advance!!!
>> >>
>> >>
>> >>
>> >> this is the the debug of one call.
>> >> Len = 73 [ 97 96 3f 85 e5 09 71 f2 5f 00 01 00 60 01 0a 00 02 08 06 83
>> >> 90 3b 38 87 0f 0a 07 02 13 90 17 02 10 86 08 01 00 1d 03 80 90 a3 31
>> >> 02 00 64 3a 06 44 05 95 00 00 00 3f 08 04 93 95 95 17 02 00 87 39 06
>> >> 31 d0 3a d0 3f c0 00 ]
>> >> FSN: 22 FIB 1
>> >> BSN: 23 BIB 1
>> >> <[1] MSU
>> >> [ 97 96 3f ]
>> >> Network Indicator: 2 Priority: 0 User Part: ISUP (5)
>> >> [ 85 ]
>> >> OPC XXXX DPC XXXX SLS 15
>> >> [ e5 09 71 f2 ]
>> >> CIC: 95
>> >> [ 5f 00 ]
>> >> Message Type: IAM
>> >> [ 01 ]
>> >> --FIXED LENGTH PARMS[4]--
>> >> Nature of Connection Indicator:
>> >> Satellites in connection: 0
>> >> Continuity Check: Check not required (0)
>> >> Outgoing half echo control device: not
>> >> included (0)
>> >> [ 00 ]
>> >> Forward Call Indicators:
>> >> Nat/Intl Call Ind: call to be treated as a
>> >> national call (0)
>> >> End to End Method Ind: no end-to-end method(s)
>> >> available (0)
>> >> Interworking Ind: no
>> >> interworking encountered (0)
>> >> End to End Info Ind: no end-to-end information
>> >> available (0)
>> >> ISDN User Part Ind: ISDN user part used all
>> >> the way (1)
>> >> ISDN User Part Pref Ind: ISDN
>> >> user part not preferred all the way (1)
>> >> ISDN Access Ind: originating access ISDN (1)
>> >> SCCP Method Ind: no indication (0)
>> >> [ 60 01 ]
>> >> Calling Party's Category:
>> >> Category: Ordinary calling subscriber (10)
>> >> [ 0a ]
>> >> Transmission Medium Requirements:
>> >> Speech (0)
>> >> [ 00 ]
>> >> --VARIABLE LENGTH PARMS[1]--
>> >> Called Party Number:
>> >> Nature of address: 3
>> >> NI: 1
>> >> Numbering plan: 1
>> >> Address signals:
>> >> [ 06 83 90 3b 38 87 0f ]
>> >> --OPTIONAL PARMS--
>> >> Calling Party Number:
>> >> Nature of address: 2
>> >> NI: 0
>> >> Numbering plan: 1
>> >> Presentation: 0
>> >> Screening: 3
>> >> Address signals: 0971200199
>> >> [ 0a 07 02 13 90 17 02 10 86 ]
>> >> Optional forward call indicator:
>> >> [ 08 01 00 ]
>> >> User Service Information:
>> >> [ 1d 03 80 90 a3 ]
>> >> Propagation Delay Counter:
>> >> Delay: 0ms
>> >> [ 31 02 00 64 ]
>> >> Unknown Parameter (0x3a):
>> >> [ 44 05 95 00 00 00 ]
>> >> Location Number:
>> >> [ 3f 08 04 93 95 95 17 02 00 87 ]
>> >> Parameter Compatibility Information:
>> >> [ 39 06 31 d0 3a d0 3f c0 ]
>> >>
>> >> Unhandled optional parameter 0x8 'Optional forward call indicator'
>> >> [0x0 ]
>> >> Unhandled optional parameter 0x31 'Propagation Delay Counter'
>> >> [0x0 0x64 ]
>> >> Unhandled optional parameter 0x3a 'Unknown'
>> >> [0x44 0x5 0x95 0x0 0x0 0x0 ]
>> >> Unhandled optional parameter 0x3f 'Location Number'
>> >> 0x4 0x93 0x95 0x95 0x17 0x2 0x0 0x87 ]
>> >> Unhandled optional parameter 0x39 'Parameter Compatibility
>> >> Information'
>> >> [0x31 0xd0 0x3a 0xd0 0x3f 0xc0 ]
>> >> Len = 16 [ 96 98 0d 85 c4 49 79 f2 5f 00 0c 02 00 02 81 81 ]
>> >> FSN: 24 FIB 1
>> >> BSN: 22 BIB 1
>> >>> [1] MSU
>> >> [ 96 98 0d
>> >>
>> >> _______________________________________________
>> >> --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
>
>
>
>
> ------------------------------
>
> Message: 6
> Date: Fri, 18 Sep 2009 09:46:16 +0200
> From: Attila Domjan <adomjan at tvnet.hu>
> Subject: Re: [asterisk-ss7] Voice is not coming in Outbound Isup Call
> To: asterisk-ss7 at lists.digium.com
> Message-ID: <1253259976.3031.5.camel at guede>
> Content-Type: text/plain; charset="us-ascii"
>
> I assume ouccered by the missing p->dialing = 0; in chan_dahdi near
> p->proceeding = 1; in case ISUP_EVENT_ACM: and case ISUP_EVENT_CPG:.
> I wrote about it in many times in this list.
>
> On Fri, 2009-09-18 at 11:41 +0530, Rajesh Mahajan wrote:
>> Hi All.
>>
>> We are using Sangoma A104u Quad Card for SS7.
>>
>> Incoming call is working fine.
>> While in outbound call is working fine but not able to hear voice on
>> the channel.
>>
>> Below is the config files
>>
>> chan_dahdi.conf
>>
>> [channels]
>> ;switchtype=euroisdn
>> usecallerid=yes
>> callwaiting=yes
>> usecallingpres=yes
>> callwaitingcallerid=yes
>> threewaycalling=yes
>> transfer=yes
>> canpark=yes
>> cancallforward=yes
>> callreturn=yes
>> echocancel=yes
>> echocancelwhenbridged=yes
>> group=1
>> callgroup=1
>> pickupgroup=1
>>
>>
>> signalling = ss7
>> ss7type = itu
>> ss7_called_nai=dynamic
>> ss7_calling_nai=dynamic
>> networkindicator=national
>>
>> ; port 1
>> linkset = 1
>> group = 1
>> signalling=ss7
>> ss7type = itu
>> context = dialout
>> pointcode = 8002
>> adjpointcode = 9146
>> defaultdpc = 9146
>> networkindicator = national
>> sigchan = 16
>> cicbeginswith = 1
>> channel => 1-15
>> cicbeginswith = 17
>> channel => 17-31
>>
>>
>> /etc/dahdi/system.conf
>>
>> loadzone=us
>> defaultzone=us
>>
>> #Sangoma A104 port 1 [slot:1 bus:12 span:1] <wanpipe1>
>> span=1,0,0,ccs,hdb3
>> bchan=1-15,17-31
>> echocanceller=mg2,1-15,17-31
>> #hardhdlc=16
>> dchan=16
>>
>> /etc/wanpipe/wanpipe1.conf
>> [devices]
>> wanpipe1 = WAN_AFT_TE1, Comment
>>
>> [interfaces]
>> w1g1 = wanpipe1, , TDM_VOICE, Comment
>>
>> [wanpipe1]
>> CARD_TYPE = AFT
>> S514CPU = A
>> CommPort = PRI
>> AUTO_PCISLOT = NO
>> PCISLOT = 1
>> PCIBUS = 12
>> FE_MEDIA = E1
>> FE_LCODE = HDB3
>> FE_FRAME = NCRC4
>> FE_LINE = 1
>> TE_CLOCK = NORMAL
>> TE_REF_CLOCK = 0
>> TE_SIG_MODE = CCS
>> TE_HIGHIMPEDANCE = NO
>> LBO = 120OH
>> FE_TXTRISTATE = NO
>> MTU = 1500
>> UDPPORT = 9000
>> TTL = 255
>> IGNORE_FRONT_END = NO
>> TDMV_SPAN = 1
>> TDMV_DCHAN = 0
>> TDMV_HW_DTMF = NO
>> TDMV_HW_FAX_DETECT = NO
>>
>> [w1g1]
>> ACTIVE_CH = ALL
>> TDMV_HWEC = NO
>>
>> _______________________________________________
>> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>>
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