[asterisk-ss7] signalling ok, but no sound

[Attila Domjan]

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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