[asterisk-ss7] Voice is not coming in Outbound Isup Call
Zoltan Markella
zoltan.markella at openhorizont.co.uk
Fri Sep 18 05:12:43 CDT 2009
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:
>
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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 --------------
>> 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
>>>
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