You are using Digium hardware? If yes then you libss7 -----Original Message----- From: asterisk-ss7-bounces@xxxxxxxxxxxxxxxx [mailto:asterisk-ss7-bounces at lists.digium.com] On Behalf Of vallimamod abdullah Sent: Friday, March 05, 2010 1:33 AM To: asterisk-ss7 at lists.digium.com Subject: No ringback tone on incoming calls with chanss7 Hello list, I am using chanss7-1.3 with asterisk-1.6.0.6 and everything works fine except that asterisk does not generate ringtone on ss7 incoming calls. I have forced option 'r' in the Dial command in my dialplan: exten => _X.,1,Dial(SIP/proxy1/${EXTEN},,r) and have set progressinband=yes in sip.conf. Here is what I get in the logs for incoming call: [2010-03-04 20:53:06.594] VERBOSE[4581] logger.c: [2010-03-04 20:53:06.594] -- Recv IAM CIC=1 ANI=***** DNI=17777xxxx RNI= redirect=no/0 complete=1 [2010-03-04 20:53:06.594] DEBUG[4581] l4isup.c: IAM cic=1, owner=0x00000000 [2010-03-04 20:53:06.594] DEBUG[4581] l4isup.c: Setting iam.dni.complete [2010-03-04 20:53:06.594] VERBOSE[4896] logger.c: [2010-03-04 20:53:06.594] -- Executing [17777xxxx at ss7:2] Dial("SS7/ls1/1", "SIP/proxy1/17777xxxx,,r") in new stack [2010-03-04 20:53:06.595] VERBOSE[4896] logger.c: [2010-03-04 20:53:06.594] == Using SIP RTP CoS mark 5 [2010-03-04 20:53:06.595] VERBOSE[4896] logger.c: [2010-03-04 20:53:06.595] -- Called proxy1/17777xxxx [2010-03-04 20:53:06.595] DEBUG[4896] l4isup.c: SS7 indicate CIC=1. [2010-03-04 20:53:06.595] DEBUG[4896] l4isup.c: Sending ALERTING call progress for CIC=1 in-band ind=0. [2010-03-04 20:53:06.595] DEBUG[4896] l4isup.c: Queue packet CIC=1, len=17, linkset='ls1', link='l1', slinkset='ls1', slink='l1' [2010-03-04 20:53:06.595] DEBUG[4581] mtp.c: Queue MSU, lsi=0, last_send_ix=0, linkset=ls1, m->link=l1 [2010-03-04 20:53:06.597] DEBUG[4581] mtp.c: Sending buffer to dahdi len=21, on link 'l1' bsn=72, fsn=117. [2010-03-04 20:53:06.838] VERBOSE[4896] logger.c: [2010-03-04 20:53:06.838] -- SIP/proxy1-b770d688 is ringing On the ss7 dump with wireshark, I have for the CPG frame: Optional backward call indicators: 0x0 0 = In-band information indicator: no indication 0 = Call diversion may occur indicator: no indication 0 = Simple segmentation indicator: no additional information will be sent 0 = MLPP user indicator: no indication Is there a way to force asterisk to generate the progress tone ? Thank you ! Regards, - vma . -- _____________________________________________________________________ -- 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