Please past your chan_dahdi.conf and system.conf. Also check if you have ulaw selected in your sip phone. On Wed, Nov 10, 2010 at 12:35 AM, dave george <dgeorge at teletoneinc.com>wrote: > I wanted to add that I checked and my CICs are lined up correctly on both > sides. I am using Asterisk 1.6.2.13. > > > > Thanks, > > Dave > > > > *From:* asterisk-ss7-bounces at lists.digium.com [mailto: > asterisk-ss7-bounces at lists.digium.com] *On Behalf Of *dave george > *Sent:* Tuesday, November 09, 2010 9:10 AM > *To:* asterisk-ss7 at lists.digium.com > *Subject:* Re: [asterisk-ss7] No Audio on SS7 calls to Remote PRIs > > > > Hi Guys, > > > > I am having a similar issue with no audio. Other end is an ericsson > switch. See the logs below. I can make and receive calls fine but no > audio. I checked my CIC and they are lined up well. I run dahdi_monitor > but I only see activity from my end (softphone registered on asterisk). > Nothing from the far end. One strange thing I notice is the # sign at the > end of the numbers in the logs. > > Example Address signals: 5339427# > > > > > > libss7 version: 1.0.2 > > DAHDI Version: 2.4.0 Echo Canceller: MG2 > > > > > > Network Indicator: 2 Priority: 0 User Part: ISUP (5) > > [ 85 ] > > OPC 2500 DPC 5352 SLS 0 > > [ e8 14 71 02 ] > > CIC: 400 > > [ 90 01 ] > > Message Type: CPG > > [ 2c ] > > --FIXED LENGTH PARMS[1]-- > > Event Information: > > ALERTING > > [ 01 ] > > --OPTIONAL PARMS-- > > Backward Call Indicator: > > Charge indicator: 2 > > Called party's status indicator: 1 > > Called party's category indicator: 1 > > End to End method indicator: 0 > > Interworking indicator: 0 > > End to End information indicator: 0 > > ISDN user part indicator: 1 > > Holding indicator: 0 > > ISDN access indicator: 1 > > Echo control device indicator: 1 > > SCCP method indicator: 0 > > [ 11 02 16 34 ] > > Optional Backward Call Indicator: > > In-band information indicator: 1 > > Call diversion may occur indicator: 1 > > Simple segmentation indicator: 0 > > MLPP user indicator: 0 > > [ 29 01 03 ] > > > > -- DAHDI/63-1 is ringing > > Len = 25 [ ba 9c 16 85 e8 14 71 02 90 01 09 01 11 02 04 34 2d 02 00 5a 39 > 02 2d c0 00 ] > > FSN: 28 FIB 1 > > BSN: 58 BIB 1 > > <[1] MSU > > [ ba 9c 16 ] > > Network Indicator: 2 Priority: 0 User Part: ISUP (5) > > [ 85 ] > > OPC 2500 DPC 5352 SLS 0 > > [ e8 14 71 02 ] > > CIC: 400 > > [ 90 01 ] > > Message Type: ANM > > [ 09 ] > > --OPTIONAL PARMS-- > > Backward Call Indicator: > > Charge indicator: 0 > > Called party's status indicator: 1 > > Called party's category indicator: 0 > > End to End method indicator: 0 > > Interworking indicator: 0 > > End to End information indicator: 0 > > ISDN user part indicator: 1 > > Holding indicator: 0 > > ISDN access indicator: 1 > > Echo control device indicator: 1 > > SCCP method indicator: 0 > > [ 11 02 04 34 ] > > Unknown Parameter (0x2d): > > [ 00 5a ] > > Parameter Compatibility Information: > > [ 39 02 2d c0 ] > > > > Unhandled optional parameter 0x2d 'Unknown' > > [0x0 0x5a ] > > Unhandled optional parameter 0x39 'Parameter Compatibility Information' > > [0x2d 0xc0 ] > > -- DAHDI/63-1 answered SIP/4735211000-00000036 > > > > > > localhost*CLI> localhost*CLI> > > > > > > #* > ( # = Audio Level * = Max Audio Hit ) > > <----------------(RX)----------------> > <----------------(TX)----------------> > > ################### > * Rx: 16 ( 16) Tx: 4570 ( 4850) > > > ########################* Rx: 16 ( 16) Tx: 1447 ( 1474) > > > #* Rx: 16 ( 16) Tx: 98 ( 98) > > > > > > Thanks, > > Dave > > > > -- > _____________________________________________________________________ > -- 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 > -- Regards, Abdul Basit | +92 32 1416 4196 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-ss7/attachments/20101110/e747cd57/attachment-0001.htm
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