[asterisk-ss7] No Audio on SS7 calls to Remote PRIs

Stephan Ellis stephan.ellis at gmail.com
Mon Oct 4 12:51:39 CDT 2010


Anyone have any ideas on this?

-stephan

On Thu, Sep 30, 2010 at 10:30 AM, Stephan Ellis <stephan.ellis at gmail.com>wrote:

> Yes that's correct.  Sorry, I should be more clear about my setup.  I work
> for a rural telephone company.  We have our asterisk box connected to a
> Siemens EWSD.  I have my softphone connected directly to the asterisk box.
> The box I am calling is an asterisk box connected to a PRI from bell.  I get
> no audio there.  I also tested against a call center that has PRIs from bell
> and I get the same issue.  Your guess is as good as mine as to what they are
> using.
>
> To complicate matters, I also have my main phone system (asterisk)
> connected to a PRI on my EWSD.  This is a completely different box, but
> connected to the same switch.  When I call it from my ss7 box I get audio
> just fine.
>
> We contacted our siemens guys about this and they say that when I call a
> remote PRI from our ss7 box, our switch is sending the asterisk box a pass
> along message, which we seem to be ignoring.
>
> Hope that clears up my situation a little better.  Thanks!
>
> -stephan
>
>
> On Thu, Sep 30, 2010 at 10:21 AM, Jean Cérien <cerien.jean at gmail.com>wrote:
>
>>
>> just to clarify... you have the following setup: ss7 -> asterisk -> sip ->
>> softphone
>>
>> where is the PRI ?
>>
>>
>>
>> On Thu, Sep 30, 2010 at 11:04 AM, Stephan Ellis <stephan.ellis at gmail.com>wrote:
>>
>>> I do see audio being received, but I don't hear it on my softphone.  I
>>> see no TX at all.  Interestingly, the guy on the pri I was calling said he
>>> could hear me.  The remote pri is an asterisk box, so i set a DID on it to
>>> go straight to the echo test.  While that system is playing demo-echo I see
>>> RX on my end, but when the actual echo test starts i see nothing.
>>>
>>> -stephan
>>>
>>>
>>> On Thu, Sep 30, 2010 at 9:45 AM, Jean Cérien <cerien.jean at gmail.com>wrote:
>>>
>>>>
>>>> Hi
>>>>
>>>> Have you tried using dahdi_monitor to see if any sound is received ?
>>>>
>>>> Rgds,
>>>> J.
>>>>
>>>>   On Thu, Sep 30, 2010 at 10:15 AM, Stephan Ellis <
>>>> stephan.ellis at gmail.com> wrote:
>>>>
>>>>>  All,
>>>>>
>>>>>   I've got a problem on my SS7 implementation.  When I originate calls
>>>>> across my SS7 link and the call lands on a PRI, I get no audio in either
>>>>> direction.  The stack I am using is:
>>>>>
>>>>> Asterisk 1.6.2.13
>>>>> DAHDI 2.4.0
>>>>> libss7 1.0.2
>>>>> libpri 1.4.11 (not sure if i need that, but thought it might be needed
>>>>> for ISUP stuff)
>>>>> WANPIPE 3.5.15.4
>>>>> Linux Kernel 2.6.18-194.11.4.el5 on Centos 5.5
>>>>>
>>>>> The whole stack was hand compiled on the server (not from repos).
>>>>>
>>>>> My dialplan is pretty simple, possibly too simple:
>>>>>
>>>>> exten => _XXXXXXX,1,Dial(DAHDI/g0/${EXTEN})
>>>>> exten => _XXXXXXX,n,Hangup()
>>>>>
>>>>> My chan_dahdi.conf looks like this:
>>>>>
>>>>> ;autogenerated by /usr/sbin/wancfg_dahdi do not hand edit
>>>>> ;autogenrated on 2010-09-24
>>>>> ;Dahdi Channels Configurations
>>>>> ;For detailed Dahdi options, view /etc/asterisk/chan_dahdi.conf.bak
>>>>>
>>>>> [trunkgroups]
>>>>>
>>>>> [channels]
>>>>> context=default
>>>>> usecallerid=yes
>>>>> hidecallerid=no
>>>>> callwaiting=yes
>>>>> usecallingpres=yes
>>>>> callwaitingcallerid=yes
>>>>> threewaycalling=yes
>>>>> transfer=yes
>>>>> canpark=yes
>>>>> cancallforward=yes
>>>>> callreturn=yes
>>>>> echocancel=no
>>>>> echocancelwhenbridged=no
>>>>> relaxdtmf=yes
>>>>> rxgain=0.0
>>>>> txgain=0.0
>>>>> group=1
>>>>> callgroup=1
>>>>> pickupgroup=1
>>>>> immediate=no
>>>>>
>>>>> ss7type=ansi
>>>>> signalling=ss7
>>>>> ss7_called_nai=dynamic
>>>>> ss7_calling_nai=dynamic
>>>>> ss7_internationalprefix=00
>>>>> ss7_nationalprefix=0
>>>>> ss7_subscriberprefix=
>>>>> ss7_unknownprefix=
>>>>> networkindicator=national
>>>>> explicitacm=yes
>>>>> linkset=1
>>>>> pointcode=1-1-1
>>>>> defaultdpc=5-9-192
>>>>> adjpointcode=5-9-192
>>>>> group=0
>>>>> cicbeginswith=1
>>>>> channel=2-24
>>>>> sigchan=1
>>>>>
>>>>> context => from-pstn
>>>>>
>>>>>
>>>>> --
>>>>> _____________________________________________________________________
>>>>> -- 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
>>>
>>
>>
>> --
>> _____________________________________________________________________
>> -- 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 --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-ss7/attachments/20101004/1184a2ef/attachment.htm 


More information about the asterisk-ss7 mailing list