[asterisk-ss7] signalling ok, but no sound

[Attila Domjan]

Yes it is needed my libss7 version. 
This chan_dahdi.c tested and compile with asterisk-1.6.0.9.

svn co https://observer.router.hu/repos_pub/libss7/trunk libss7
svn co https://observer.router.hu/repos_pub/chan_dahdi/trunk chan_dahdi

but it needed many additional configuration options in chan_dahdi.conf
for properly operation (timers, etc). If you decide to try it I can help
you with a sample configuration.

A

On Fri, 2009-06-05 at 12:55 +0100, Zoltan Markella wrote:
> Attila,
> 
> You're a genius! Thanks for the quick reply - it was indeed the problem.
> I have tried to grab chan_dahdi from your svn (and replace the stock 
> one), but I had quite a few compilation errors, so I abandoned that 
> route. (Do I need to get you libss7 as well to make it work btw?)
> 
> I've seen this p->dialing = 0 problem in the list archives, but I wasn't 
> sure where it was missing from.
> 
> Anyway, thanks again for the help!
> 
> Cheers,
> Zoltan
> 
> Attila Domjan wrote:
> > I think it is the bug in the chan_dahdi, which is introduced by the
> > p->dialing not implemented proberly in the ss7 part of the chan_dahdi.
> >
> > Check wheter exists p->dialing = 0; after the p->progress = 1;
> > in static void *ss7_linkset(void *data) function at the
> >
> > case CPG_EVENT_INBANDINFO:
> > case ISUP_EVENT_ACM:
> >
> >
> > On Fri, 2009-06-05 at 11:24 +0100, Zoltan Markella wrote:
> >   
> >> Hi,
> >>
> >> I've been trying to get SS7 working with asterisk the last couple of 
> >> days, but had on luck.
> >>
> >> My configuration:
> >> - server with a Digium TE420 card, another server with a Digium TE120 
> >> card crossover cable between
> >> - libss7-1.0.2
> >> - dahdi-2.1.0.4
> >> - asterisk-1.6.1.1
> >>
> >> The connection between the two servers is working fine. I have set up a 
> >> test pri_cpe/pri_net signalling and was able to do a SIP->DAHDI->SIP call.
> >>
> >> /etc/dahdi/system.conf (both machines):
> >> span=1,1,0,ccs,hdb3,crc4
> >> bchan=1-15,17-31
> >> mtp2=16
> >> chocanceller=mg2,1-15,17-31
> >>
> >> /etc/asterisk/chan_dahdi.conf (server1)
> >> context=from-ss7
> >> signalling = ss7
> >> ss7type = itu
> >> linkset = 1
> >> pointcode = 20
> >> adjpointcode = 25
> >> defaultdpc = 25
> >> ss7_called_nai=dynamic
> >> ss7_calling_nai=dynamic
> >> networkindicator=international
> >>
> >> cicbeginswith = 1
> >> channel = 1-15
> >> cicbeginswith = 17
> >> channel = 17-31
> >> sigchan = 16
> >>
> >> /etc/asterisk/chan_dahdi.conf (server2)
> >> pointcode = 25
> >> adjpointcode = 20
> >> defaultdpc = 20
> >> [otherwise same as server1's config]
> >>
> >> After starting up both server, SS7 comes up successfully:
> >> MTP2 link up (SLC 0)
> >> --- SS7 Up ---
> >> Resetting CICs 1 to 15
> >> Resetting CICs 17 to 31
> >> Got reset acknowledgement from CIC 1 to 15.
> >> Got reset acknowledgement from CIC 17 to 31.
> >>
> >> Here's my call scenario:
> >> SIP/600 (Grandstream phone) -> server 1 -> SS7 -> server 2 -> SIP/555 
> >> (Nokia E71)
> >>
> >> Output from server 1:
> >>   == Using SIP RTP CoS mark 5
> >>     -- Executing [1000 at default:1] Dial("SIP/600-08887770", 
> >> "DAHDI/g1/1000,55,tTo") in new stack
> >>     -- Called g1/1000
> >>     -- DAHDI/1-1 is proceeding passing it to SIP/600-08887770
> >>     -- DAHDI/1-1 is ringing
> >>     -- DAHDI/1-1 answered SIP/600-08887770
> >>
> >> Output from server 2:
> >> -- Executing [1000 at from-ss7:1] Ringing("DAHDI/1-1", "") in new stack
> >>     -- Accepting call to '1000' on CIC 1
> >>     -- Executing [1000 at from-ss7:2] Dial("DAHDI/1-1", "SIP/555,50,tTo") 
> >> in new stack
> >>   == Using SIP RTP CoS mark 5
> >>     -- Called 555
> >>     -- SIP/555-0890bab0 is ringing
> >>     -- SIP/555-0890bab0 answered DAHDI/1-1
> >>
> >> So the call is set up properly. BUT there's no audio on either end!!!
> >> With dahdi_monitor I can see activity on both card's first channel (and 
> >> no other channels, so there's no CIC mismatch), but on server1 I only 
> >> have RX and on server2 I only have TX.
> >>
> >> Could anybody give me a hint where my problem could lie?
> >>
> >> Cheers,
> >> Zoltan
> >>
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