I got this working properly. The basic problem was CIC lineup/mapping. Our mux had the E1s mapped incorrectly. So what we thought was E1-1 was E1-5. Thanks, Dave From: asterisk-ss7-bounces@xxxxxxxxxxxxxxxx [mailto:asterisk-ss7-bounces at lists.digium.com] On Behalf Of Abdul Basit Sent: Tuesday, November 09, 2010 3:41 PM To: asterisk-ss7 at lists.digium.com Subject: Re: No Audio on SS7 calls to Remote PRIs 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@xxxxxxxxxxxxxxxx [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: 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/d31e2ed3/attachment-0001.htm