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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