Hi, I have a problem with chan_ss7 & Asterisk. If an IP Phone (SIP) calls a PSTN number and the call is terminated by the calling party chan_ss7 won't (?) recieve a release cause from Asterisk. In my case this means that the release cause will be 41 (Temporary Failure) in the connecting EWSD switch instead of 16 (Normal Call Clearing). IP phone calls PSTN -> IP side releases call -> Temporary failure IP phone calls PSTN -> PSTN side releases call -> Normal Call Clearing PSTN calls IP phone -> IP side releases call -> Normal Call Clearing PSTN calls IP phone -> PSTN side releases call -> Normal Call Clearing I tried with: Asterisk 1.2.10 & chan_ss7 0.8.4 Asterisk 1.2.10 & chan_ss7 0.9 Asterisk 1.2.13 & chan_ss7 0.9 Based on chan_ss7 0.9 l4isup.c, I guess the developers are aware of this poblem: static void initiate_release_circuit(struct ss7_chan* pvt, int cause) { pvt->hangupcause = cause; /* Remember for REL retransmit */ /* We sometimes get hangupcause=0 (seen when no match in dialplan, not even invalid handler). This doesn't work too well, for example ast_softhangup() doesn't actually hang up when hangupcause=0. */ if(pvt->hangupcause == 0) { pvt->hangupcause = AST_CAUSE_NORMAL_TEMPORARY_FAILURE; } isup_send_rel(pvt, pvt->hangupcause); pvt->state = ST_SENT_REL; /* Set up timer T1 and T5 waiting for RLC. */ t1_start(pvt); t5_start(pvt); } Anybody knows a way to correct this? Preferably within Asterisk. As a temporal solution I modified l4isup.c, but I think it's not the correct place to do it. Regards, Peter -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-ss7/attachments/20061122/a9fcdabf/attachment.htm