How to solve this problem ? On Fri, Sep 18, 2009 at 1:16 PM, <asterisk-ss7-requ...@lists.digium.com> wrote: > Send asterisk-ss7 mailing list submissions to > asterisk-...@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-requ...@lists.digium.com > > You can reach the person managing the list at > asterisk-ss7-ow...@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.raf...@gmail.com> > Subject: Re: [asterisk-ss7] handling * and # of dialed number on the > extension.conf > To: asterisk-ss7@lists.digium.com > Message-ID: > <b1b91df00909171132q6d20a908if4b012c703f5c...@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 <gustavomars...@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 <originaljimda...@gmail.com> > Subject: [asterisk-ss7] SS7 for Verisign A-Link, M3UA? > To: asterisk-ss7@lists.digium.com > Message-ID: > <cb0ab51a0909171542j24e6fba1j8bf6f5c399b38...@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: originaljimda...@gmail.com > AIM: originaljimdandy > > > > ------------------------------ > > Message: 3 > Date: Fri, 18 Sep 2009 11:41:52 +0530 > From: Rajesh Mahajan <rajeshmahaja...@gmail.com> > Subject: [asterisk-ss7] Voice is not coming in Outbound Isup Call > To: asterisk-ss7@lists.digium.com > Message-ID: > <c9961d450909172311o3c36da4wcd51b0580242d...@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 <wa...@convergence.pk> > Subject: Re: [asterisk-ss7] Voice is not coming in Outbound Isup Call > To: asterisk-ss7@lists.digium.com > Message-ID: > <b8ad2a5b0909172319j2de6f2e1p9eb75b2fca5b6...@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 > <rajeshmahaja...@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" <kkovac...@varna.net> > Subject: Re: [asterisk-ss7] handling * and # of dialed number on the > extension.conf > To: asterisk-ss7@lists.digium.com > Message-ID: <20090918064231.m36...@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 <gustavomars...@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 <adom...@tvnet.hu> > Subject: Re: [asterisk-ss7] Voice is not coming in Outbound Isup Call > To: asterisk-ss7@lists.digium.com > Message-ID: <1253259976.3031.5.ca...@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 > > ------------------------------ > > 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