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 C?rien <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 C?rien <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/20100930/83be7628/attachment.htm
|