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 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... 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