[asterisk-ss7] Link Up but no audio
Matthew Fredrickson
creslin at digium.com
Wed Apr 22 12:20:54 CDT 2009
Daniel Pizarro Bustamante wrote:
> Hello all,
>
> Finally, I had to downgrade the asterisk/zaptel and use chan_ss7. The linkset is up and I can hear audio in both directions.
>
> I think there was something wrong in the libss7-1.0 with the asterisk-trunk because the link had those symptoms:
> - Originating a call from CLI with Originate command:
> - application SendDTMF works, and with dahdi_monitor I could see TX and RX traffic (when I talk).
> - application Playback doesn't work, and with dahdi_monitor I could not see TX traffic (0 bytes). I could see RX traffic (when I talk).
> - I use Monitor to record the call, and the file-in (RX) is empty, the file-out (TX) has the audio of the played prompt.
>
> I hope it could be helpful for someone.
I'm fairly certain that the issue was with CIC mapping then. If your
CICs were mapped correctly, and you're talking in both directions in the
call (which you should be in this test), you should see audio in both
directions, TX and RX. In cases where you have CIC mismatch, you will
see audio in only one direction (either TX or RX depending on whether
you are on the right or wrong channel).
Matt
>
> Regards,
>
> Daniel
>
> --- El lun 20-abr-09, Daniel Pizarro Bustamante <dpizarro at yahoo.com> escribió:
>
>> De: Daniel Pizarro Bustamante <dpizarro at yahoo.com>
>> Asunto: Re: [asterisk-ss7] Link Up but no audio
>> A: asterisk-ss7 at lists.digium.com
>> Fecha: lunes, 20 abril, 2009, 6:39 pm
>>
>> Hello,
>>
>> When I use SendDTMF the operator and the dahdi_monitor
>> "see" the TX stream. It seems there is something wrong
>> between Asterisk and the libss7.
>>
>> Please help!
>>
>> Daniel
>>
>>
>>
>> --- El sáb 18-abr-09, Daniel Pizarro Bustamante <dpizarro at yahoo.com>
>> escribió:
>>
>>> De: Daniel Pizarro Bustamante <dpizarro at yahoo.com>
>>> Asunto: Re: [asterisk-ss7] Link Up but no audio
>>> A: asterisk-ss7 at lists.digium.com
>>> Fecha: sábado, 18 abril, 2009, 6:17 pm
>>>
>>> Hello Matthew:
>>>
>>> I send an call by channel number 1, and I see the
>> audio
>>> only in one-way (RX).
>>>
>>> [root at pelxsms2voice ~]# dahdi_monitor 1 -vv
>>>
>>> Visual Audio Levels.
>>> --------------------
>>> Use chan_dahdi.conf file to adjust the gains if
>> needed.
>>> ( # = Audio Level * = Max Audio Hit )
>>> <----------------(RX)---------------->
>>> ###################################*
>>>
>>>
>>>
>>> <----------------(TX)---------------->
>>> Rx: 11893 (15967) Tx: 0 (
>>> 0)
>>>
>>> So I think, the CIC mismatch is not the issue.
>>>
>>> Daniel
>>>
>>>
>>>
>>> --- El sáb 18-abr-09, Matthew Fredrickson <creslin at digium.com>
>>> escribió:
>>>
>>>> De: Matthew Fredrickson <creslin at digium.com>
>>>> Asunto: Re: [asterisk-ss7] Link Up but no audio
>>>> A: asterisk-ss7 at lists.digium.com
>>>> Fecha: sábado, 18 abril, 2009, 1:34 pm
>>>> Daniel Pizarro Bustamante wrote:
>>>>> Hello Mesbah:
>>>>>
>>>>> I am using the libss7-1.0 because the
>>> libss7-trunk has
>>>> an error that we could see in this forum.
>>>>> The asterisk and dahdi was downloaded from
>> the
>>> svn
>>>> (trunk versions).
>>>>> How can I detect the CIC mismatch?
>>>> Sorry, hit the wrong button on my last email.
>>>>
>>>> That was meant to say:
>>>>
>>>> One way to do it is to bring a call up, run a
>>> playback, and
>>>> use
>>>> dahdi_monitor -v on the channel you think it
>> should
>>> be
>>>> on. If you do
>>>> not see the audio levels present on TX or RX, you
>> have
>>> the
>>>> CIC wrong.
>>>>
>>>> You can then run dahdi_monitor -v on the
>> surrounding
>>>> channels to see if
>>>> you're off by one or two on that channel.
>>>>
>>>> Many people don't configure the subsequent spans
>>> correctly
>>>> after the
>>>> first one. So, for example, for an E1,
>> oftentimes a
>>>> telco will
>>>> configure the first span with 32 CICs, even
>> though
>>> DAHDI
>>>> registers only
>>>> 31 physical channels, and people forget to add a
>> new
>>>> "cicbeginswith=33"
>>>> for the channels in the second span, causing an
>> off by
>>> one
>>>> CIC error
>>>> starting with the second span (and each
>> subsequent
>>> span).
>>>> Matthew Fredrickson
>>>> Digium, Inc.
>>>>
>>>>> Thanks,
>>>>>
>>>>> Daniel Pizarro
>>>>>
>>>>>
>>>>> Connected to Asterisk SVN-
>>>>>
>>>>>
>>>>> --- El sáb 18-abr-09, Mesbahuddin Malik
>> <mesbah.malik at gmail.com>
>>>> escribió:
>>>>>> De: Mesbahuddin Malik <mesbah.malik at gmail.com>
>>>>>> Asunto: Re: [asterisk-ss7] Link Up but
>> no
>>> audio
>>>>>> A: asterisk-ss7 at lists.digium.com
>>>>>> Fecha: sábado, 18 abril, 2009, 11:40
>> am
>>>>>> Hi,
>>>>>>
>>>>>> For various reasons this happen. The
>> main
>>> reasons
>>>> are
>>>>>>
>>>>>> 1.Firewall
>>>>>>
>>>>>> 2.Libss7 version Mitch Matching
>>>>>>
>>>>>> 3. CIC Mismatch
>>>>>>
>>>>>> 4. Terminal End firewall issues
>>>>>>
>>>>>>
>>>>>> so check the above and let me know . I
>> can
>>> help
>>>> you in
>>>>>> this respect.
>>>>>>
>>>>>> Best regards
>>>>>> Mesbah
>>>>>>
>>>>>>
>>>>>> On Sat, Apr 18, 2009 at 8:14 PM,
>>>>>> Daniel Pizarro Bustamante <dpizarro at yahoo.com>
>>>>>> wrote:
>>>>>>
>>>>>>
>>>>>>
>>>>>> Hello:
>>>>>>
>>>>>> I have an SS7 deployment with Asterisk
>>>>>> 1.6.x/libss7(1.0.x)/DAHDI.
>>>>>>
>>>>>> I try to originate a call (from CLI) and
>> I am
>>> able
>>>> to
>>>>>> connect the call
>>>>>>
>>>>>> but the called party doesn't hear
>> anything.
>>> Eg.
>>>>>> Originate Local/01996613358 at llamada
>>> application
>>>> Playback
>>>>>> tt-monkeys
>>>>>>
>>>>>> I can send dtmf's and the called party
>> can
>>> hear
>>>> them.
>>>>>> Originate Local/01996613358 at llamada
>>> application
>>>> SendDTMF
>>>>>> 12345678
>>>>>>
>>>>>>
>>>>>> The debug shows me that:
>>>>>>
>>>>>> -- Executing
>>>> [01996613358 at llamada:1]
>>>>>> Set("Local/01996613358 at llamada-1bb8;2",
>>>>>> "CALLERID(number)=7773") in new
>>>>>> stack
>>>>>> -- Executing
>>>> [01996613358 at llamada:2]
>>>>>>
>> Dial("Local/01996613358 at llamada-1bb8;2",
>>>>>> "DAHDI/g1/01996613358") in new
>>>>>> stack
>>>>>> -- Called g1/01996613358
>>>>>> Len = 34 [ b2 af 1f c5 38 38 34 1e 01 00
>> 01
>>> 00 60
>>>> 01 0a 00
>>>>>> 02 0a 08 03
>>>>>> 10 10 99 66 31 53 f8 0a 04 04 10 77 37
>> 00 ]
>>>>>> FSN: 47 FIB 1
>>>>>> BSN: 50 BIB 1
>>>>>>> [0] MSU
>>>>>> [ b2 af 1f ]
>>>>>> Network Indicator:
>>>> 3 Priority: 0 User Part: ISUP
>>>>>> (5)
>>>>>> [ c5 ]
>>>>>> OPC 14544 DPC 14392
>>>> SLS 1
>>>>>> [ 38 38 34 1e ]
>>>>>>
>>>> CIC: 1
>>>>>>
>>>> [ 01 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 ]
>>>>>> pelxsms2voice*CLTransmission Medium
>>> Requirements:
>>>>>>
>>>> Speech (0)
>>>>>>
>>>> [ 00 ]
>>>>>>
>>>> --VARIABLE LENGTH PARMS[1]--
>>>>>>
>>>> Called Party Number:
>>>>>>
>>>> Nature of address: 3
>>>>>>
>>>> NI: 0
>>>>>>
>>>> Numbering plan: 1
>>>>>>
>>>> Address signals:
>>>>>> 01996613358#
>>>>>>
>>>> [ 08 03 10 10 99 66
>>>> 31 53
>>>>>> f8 ]
>>>>>>
>>>> --OPTIONAL PARMS--
>>>>>>
>>>> Calling Party Number:
>>>>>>
>>>> Nature of address: 4
>>>>>>
>>>> NI: 0
>>>>>>
>>>> Numbering plan: 1
>>>>>>
>>>> Presentation: 0
>>>>>>
>>>> Screening: 0
>>>>>>
>>>> Address signals:
>>>> 7773
>>>>>>
>>>> [ 0a 04 04 10 77 37
>>>> ]
>>>>>> Len = 18 [ af b3 0f c5 d0 38 0e ae 01 00
>> 06
>>> 12 34
>>>> 01 29 01
>>>>>> 03 00 ]
>>>>>> FSN: 51 FIB 1
>>>>>> BSN: 47 BIB 1
>>>>>> <[0] MSU
>>>>>> [ af b3 0f ]
>>>>>>
>>>>>> Network Indicator:
>>>> 3 Priority: 0 User Part: ISUP
>>>>>> (5)
>>>>>> [ c5 ]
>>>>>> OPC 14392 DPC 14544
>>>> SLS 10
>>>>>> [ d0 38 0e ae ]
>>>>>>
>>>> CIC: 1
>>>>>>
>>>> [ 01 00 ]
>>>>>>
>>>> Message Type: ACM
>>>>>>
>>>> [ 06 ]
>>>>>>
>>>> --FIXED LENGTH PARMS[1]--
>>>>>>
>>>> Backward Call Indicator:
>>>>>>
>>>> Charge indicator: 2
>>>>>>
>>>> Called party's
>>>> status
>>>>>> indicator: 0
>>>>>>
>>>> Called party's
>>>>>> category indicator: 1
>>>>>>
>>>>>>
>>>> End to End method
>>>>>> indicator: 0
>>>>>>
>>>> Interworking
>>>> indicator: 0
>>>>>>
>>>> End to End
>>>> information
>>>>>> indicator: 0
>>>>>>
>>>> ISDN user part
>>>> indicator:
>>>>>> 1
>>>>>>
>>>>>>
>>>> Holding indicator: 0
>>>>>>
>>>> ISDN access
>>>> indicator: 1
>>>>>>
>>>> Echo control device
>>>>>> indicator: 1
>>>>>>
>>>> SCCP method
>>>> indicator: 0
>>>>>>
>>>> [ 12 34 ]
>>>>>>
>>>> --OPTIONAL PARMS--
>>>>>>
>>>> Optional Backward Call Indicator:
>>>>>>
>>>> In-band information
>>>>>> indicator: 1
>>>>>>
>>>> Call diversion may
>>>> occur
>>>>>> indicator: 1
>>>>>>
>>>> Simple segmentation
>>>>>> indicator: 0
>>>>>>
>>>>>>
>>>> MLPP user indicator:
>>>> 0
>>>>>>
>>>> [ 29 01 03 ]
>>>>>> -- DAHDI/1-1 is proceeding
>>>> passing it to
>>>>>> Local/01996613358 at llamada-1bb8;2
>>>>>> Len = 17 [ af b4 0e c5 d0 38 0e ae 01 00
>> 09
>>> 01 11
>>>> 02 16 34
>>>>>> 00 ]
>>>>>>
>>>>>> FSN: 52 FIB 1
>>>>>> BSN: 47 BIB 1
>>>>>> <[0] MSU
>>>>>> [ af b4 0e ]
>>>>>> Network Indicator:
>>>> 3 Priority: 0 User Part: ISUP
>>>>>> (5)
>>>>>> [ c5 ]
>>>>>> OPC 14392 DPC 14544
>>>> SLS 10
>>>>>> [ d0 38 0e ae ]
>>>>>>
>>>> CIC: 1
>>>>>>
>>>> [ 01 00 ]
>>>>>>
>>>> Message Type: ANM
>>>>>>
>>>> [ 09 ]
>>>>>>
>>>> --OPTIONAL PARMS--
>>>>>>
>>>> Backward Call Indicator:
>>>>>>
>>>> Charge indicator: 2
>>>>>>
>>>> Called party's
>>>> status
>>>>>> indicator: 1
>>>>>>
>>>>>>
>>>> Called party's
>>>>>> category indicator: 1
>>>>>>
>>>> End to End method
>>>>>> indicator: 0
>>>>>>
>>>> Interworking
>>>> indicator: 0
>>>>>>
>>>> End to End
>>>> information
>>>>>> indicator: 0
>>>>>>
>>>>>>
>>>> ISDN user part
>>>> indicator:
>>>>>> 1
>>>>>>
>>>> Holding indicator: 0
>>>>>>
>>>> ISDN access
>>>> indicator: 1
>>>>>>
>>>> Echo control device
>>>>>> indicator: 1
>>>>>>
>>>> SCCP method
>>>> indicator: 0
>>>>>>
>>>> [ 11 02 16 34 ]
>>>>>> -- DAHDI/1-1 answered
>>>> Local/01996613358 at llamada-1bb8;2
>>>>>> -- Launching SendDTMF(9876543210)
>>>> on
>>>>>> Local/01996613358 at llamada-1bb8;1
>>>>>> == Spawn extension (llamada,
>> 01996613358,
>>> 2)
>>>> exited
>>>>>> non-zero on
>>>>>>
>>>>>> 'Local/01996613358 at llamada-1bb8;2'
>>>>>> Len = 16 [ b4 b0 0d c5 38 38 34 1e 01 00
>> 0c
>>> 02 00
>>>> 02 81 90
>>>>>> ]
>>>>>> FSN: 48 FIB 1
>>>>>> BSN: 52 BIB 1
>>>>>>> [0] MSU
>>>>>> -- Hungup 'DAHDI/1-1'
>>>>>> [ b4 b0 0d ]
>>>>>> Network Indicator:
>>>> 3 Priority: 0 User Part: ISUP
>>>>>> (5)
>>>>>>
>>>>>> [ c5 ]
>>>>>> OPC 14544 DPC 14392
>>>> SLS 1
>>>>>> [ 38 38 34 1e ]
>>>>>>
>>>> CIC: 1
>>>>>>
>>>> [ 01 00 ]
>>>>>>
>>>> Message Type: REL
>>>>>>
>>>> [ 0c ]
>>>>>>
>>>> --VARIABLE LENGTH PARMS[1]--
>>>>>>
>>>> Cause Indicator:
>>>>>>
>>>> Coding Standard: 0
>>>>>>
>>>> Location: 1
>>>>>>
>>>> Cause Class: 1
>>>>>>
>>>> Cause Subclass: 0
>>>>>>
>>>> Cause: Normal call
>>>>>> clearing (16)
>>>>>>
>>>>>>
>>>> [ 02 81 90 ]
>>>>>> Len = 12 [ b0 b5 09 c5 d0 38 0e ae 01 00
>> 10
>>> 00 ]
>>>>>> FSN: 53 FIB 1*CLI>
>>>>>> BSN: 48 BIB 1*CLI>
>>>>>> <[0] MSUvoice*CLI>
>>>>>> [ b0 b5 09 ]e*CLI>
>>>>>> pelxsms2Network Indicator: 3 Priority:
>> 0
>>> User
>>>> Part: ISUP
>>>>>> (5)
>>>>>>
>>>>>> pelxsms2[ c5 ]CLI>
>>>>>> pelxsms2OPC 14392 DPC 14544 SLS 10
>>>>>> pelxsms2[ d0 38 0e ae ]
>>>>>> pelxsms2voice*CLCIC: 1
>>>>>> pelxsms2voice*CL[ 01 00 ]
>>>>>> pelxsms2voice*CLMessage Type: RLC
>>>>>> pelxsms2voice*CL[ 10 ]
>>>>>> pelxsms2voice*CLI>
>>>>>>
>>>>>> Len = 18 [ b0 b6 0f c1 d0 f8 0f 0e 11 80
>> 87
>>> 88 89
>>>> 8a 8b 8c
>>>>>> 8d 8e ]
>>>>>> FSN: 54 FIB 1
>>>>>> BSN: 48 BIB 1
>>>>>> <[0] MSU
>>>>>> [ b0 b6 0f ]
>>>>>> Network Indicator:
>>>> 3 Priority: 0 User Part:
>>>>>> STD_TEST (1)
>>>>>> [ c1 ]
>>>>>> OPC 14399 DPC 14544
>>>> SLS 0
>>>>>> [ d0 f8 0f 0e ]
>>>>>> H0: 1 H1: 1
>>>>>> [ 11 ]
>>>>>>
>>>>>> Len = 18 [ b6 b1 0f c1 3f 38 34 0e 21 80
>> 87
>>> 88 89
>>>> 8a 8b 8c
>>>>>> 8d 8e ]
>>>>>> FSN: 49 FIB 1
>>>>>> BSN: 54 BIB 1
>>>>>>> [0] MSU
>>>>>> [ b6 b1 0f ]
>>>>>> Network Indicator:
>>>> 3 Priority: 0 User Part:
>>>>>> STD_TEST (1)
>>>>>>
>>>>>> [ c1 ]
>>>>>> OPC 14544 DPC 14399
>>>> SLS 0
>>>>>> [ 3f 38 34 0e ]
>>>>>> H0: 1 H1: 2
>>>>>> pelxsms2[ 21 ]CLI>
>>>>>>
>>>>>>
>>>>>> Please, give a hand with that.
>>>>>>
>>>>>>
>>>>>> --
>>>>>> Daniel Pizarro Bustamante
>>>>>>
>>>>>>
>>>>>>
>>>>>> ¡Obtén la mejor experiencia
>>>> en la web!<
>>>>>> Descarga gratis el nuevo Internet
>> Explorer 8.
>>> http://downloads.yahoo.com/ieak8/?l=e1
>>>>>>
>>>>>>
>>> _______________________________________________
>>>>>> --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
>>>>>>
>>>>>>
>>>>>>
>>>>>> -----Adjunto en línea a
>> continuación-----
>>>>>>
>>> _______________________________________________
>>>>>> --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
>>>>>
>>>>> ¡Obtén la mejor
>>>> experiencia en la web!< Descarga gratis
>> el
>>> nuevo
>>>> Internet Explorer 8. http://downloads.yahoo.com/ieak8/?l=e1
>>>>>
>> _______________________________________________
>>>>> --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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>>>>
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>>>> http://lists.digium.com/mailman/listinfo/asterisk-ss7
>>>>
>>>
>>>
>>>
>> ____________________________________________________________________________________
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>>> Descarga gratis el nuevo Internet Explorer 8.
>>> http://downloads.yahoo.com/ieak8/?l=e1
>>>
>>> _______________________________________________
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>>>
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>>>
>>
>>
>> ____________________________________________________________________________________
>> ¡Obtén la mejor experiencia en la web!
>> Descarga gratis el nuevo Internet Explorer 8.
>> http://downloads.yahoo.com/ieak8/?l=e1
>>
>> _______________________________________________
>> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>>
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>>
>
>
> ____________________________________________________________________________________
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> http://downloads.yahoo.com/ieak8/?l=e1
>
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>
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