[asterisk-ss7] Voice is not coming in Outbound Isup Call
Attila Domjan
adomjan at tvnet.hu
Fri Sep 18 05:11:30 CDT 2009
Add the missing lines (p->dialing = 0) in chan_dahdi.c if not exists
near p->proceeding = 1; at the described places.
On Fri, 2009-09-18 at 15:31 +0530, Rajesh Mahajan wrote:
> How to solve this problem ?
>
> On Fri, Sep 18, 2009 at 1:16 PM, <asterisk-ss7-request at lists.digium.com> wrote:
> > Send asterisk-ss7 mailing list submissions to
> > asterisk-ss7 at lists.digium.com
> >
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> > or, via email, send a message with subject or body 'help' to
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> >
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> >
> > When replying, please edit your Subject line so it is more specific
> > than "Re: Contents of asterisk-ss7 digest..."
> >
> >
> > 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 --------------
> > An HTML attachment was scrubbed...
> > URL: http://lists.digium.com/pipermail/asterisk-ss7/attachments/20090918/2cdf25e6/attachment-0001.htm
> >
> > ------------------------------
> >
> > 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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