Hi, Here are the instructions from Attila that helped me: In chan_dahdi.c check wheter 'p->dialing = 0;' exists after the 'p->progress = 1;' in static void *ss7_linkset(void *data) function in the following cases: case CPG_EVENT_INBANDINFO: case ISUP_EVENT_ACM: Cheers, Zoltan 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 >> ******************************************* >> >> > > _______________________________________________ > --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 >
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