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 > > > > To subscribe or unsubscribe via the World Wide Web, visit > > http://lists.digium.com/mailman/listinfo/asterisk-ss7 > > or, via email, send a message with subject or body 'help' to > > asterisk-ss7-request at lists.digium.com > > > > You can reach the person managing the list at > > asterisk-ss7-owner at lists.digium.com > > > > 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-- > >> > >> asterisk-ss7 mailing list > >> To UNSUBSCRIBE or update options visit: > >> http://lists.digium.com/mailman/listinfo/asterisk-ss7 > > -------------- next part -------------- > > A non-text attachment was scrubbed... > > Name: not available > > Type: application/pgp-signature > > Size: 189 bytes > > Desc: This is a digitally signed message part > > Url : http://lists.digium.com/pipermail/asterisk-ss7/attachments/20090918/dbfb2c36/attachment.pgp > > > > ------------------------------ > > > > _______________________________________________ > > --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 > > > > End of asterisk-ss7 Digest, Vol 55, Issue 7 > > ******************************************* > > > > 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