Please past your chan_dahdi.conf and system.conf.<div><br></div><div>Also check if you have ulaw selected in your sip phone.</div><div><br></div><div><br><br><div class="gmail_quote">On Wed, Nov 10, 2010 at 12:35 AM, dave george <span dir="ltr"><<a href="mailto:dgeorge@teletoneinc.com">dgeorge@teletoneinc.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
<div lang="EN-US" link="blue" vlink="purple">
<div>
<p class="MsoNormal"><span style="font-size:11.0pt;color:#1F497D">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.</span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;color:#1F497D"> </span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;color:#1F497D">Thanks,</span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;color:#1F497D">Dave</span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;color:#1F497D"> </span></p>
<div>
<div style="border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in 0in 0in">
<p class="MsoNormal"><b><span style="font-size:10.0pt">From:</span></b><span style="font-size:10.0pt">
<a href="mailto:asterisk-ss7-bounces@lists.digium.com" target="_blank">asterisk-ss7-bounces@lists.digium.com</a>
[mailto:<a href="mailto:asterisk-ss7-bounces@lists.digium.com" target="_blank">asterisk-ss7-bounces@lists.digium.com</a>] <b>On Behalf Of </b>dave george<br>
<b>Sent:</b> Tuesday, November 09, 2010 9:10 AM<br>
<b>To:</b> <a href="mailto:asterisk-ss7@lists.digium.com" target="_blank">asterisk-ss7@lists.digium.com</a><br>
<b>Subject:</b> Re: [asterisk-ss7] No Audio on SS7 calls to Remote PRIs</span></p>
</div>
</div><div><div></div><div class="h5">
<p class="MsoNormal"> </p>
<p class="MsoNormal"><span style="font-size:11.0pt;color:#1F497D">Hi Guys,</span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;color:#1F497D"> </span></p>
<p><span style="font-size:11.0pt;color:#1F497D">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.</span></p>
<p><span style="font-size:11.0pt;color:#1F497D">Example </span>Address signals: 5339427#</p>
<p class="MsoNormal"><span style="font-size:11.0pt;color:#1F497D"> </span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;color:#1F497D"> </span></p>
<p>libss7 version: 1.0.2</p>
<p>DAHDI Version: 2.4.0 Echo Canceller: MG2</p>
<p> </p>
<p><span style="color:#1F497D"> </span></p>
<p> Network Indicator:
2 Priority: 0 User Part: ISUP (5)</p>
<p> [ 85 ]</p>
<p> OPC 2500 DPC
5352 SLS 0</p>
<p> [ e8 14 71 02
]</p>
<p>
CIC: 400</p>
<p>
[ 90 01 ]</p>
<p>
Message Type: CPG</p>
<p>
[ 2c ]</p>
<p>
--FIXED LENGTH PARMS[1]--</p>
<p>
Event
Information:</p>
<p>
ALERTING</p>
<p>
[ 01 ]</p>
<p>
--OPTIONAL PARMS--</p>
<p>
Backward Call Indicator:</p>
<p>
Charge indicator: 2</p>
<p>
Called party's status indicator: 1</p>
<p>
Called party's category indicator: 1</p>
<p>
End to End method indicator: 0</p>
<p>
Interworking indicator: 0</p>
<p>
End to End information indicator: 0</p>
<p>
ISDN user part indicator: 1</p>
<p>
Holding indicator: 0</p>
<p>
ISDN access indicator: 1</p>
<p>
Echo control device indicator: 1</p>
<p>
SCCP method indicator: 0</p>
<p>
[
11 02 16 34 ]</p>
<p>
Optional Backward Call Indicator:</p>
<p>
In-band information indicator: 1</p>
<p>
Call diversion may occur indicator: 1</p>
<p>
Simple segmentation indicator: 0</p>
<p>
MLPP user indicator: 0</p>
<p>
[ 29 01 03 ]</p>
<p> </p>
<p> -- DAHDI/63-1 is ringing</p>
<p>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 ]</p>
<p>FSN: 28 FIB 1</p>
<p>BSN: 58 BIB 1</p>
<p><[1] MSU</p>
<p>[ ba 9c 16 ]</p>
<p> Network
Indicator: 2 Priority: 0 User Part: ISUP (5)</p>
<p> [ 85 ]</p>
<p> OPC 2500 DPC
5352 SLS 0</p>
<p> [ e8 14 71 02
]</p>
<p>
CIC: 400</p>
<p>
[ 90 01 ]</p>
<p>
Message Type: ANM</p>
<p>
[ 09 ]</p>
<p>
--OPTIONAL PARMS--</p>
<p>
Backward Call Indicator:</p>
<p>
Charge indicator: 0</p>
<p>
Called party's status indicator: 1</p>
<p>
Called party's category indicator: 0</p>
<p>
End to End method indicator: 0</p>
<p>
Interworking indicator: 0</p>
<p>
End to End information indicator: 0</p>
<p>
ISDN user part indicator: 1</p>
<p>
Holding indicator: 0</p>
<p>
ISDN
access indicator: 1</p>
<p>
Echo control device indicator: 1</p>
<p>
SCCP method indicator: 0</p>
<p>
[ 11 02 04 34 ]</p>
<p>
Unknown Parameter (0x2d):</p>
<p>
[ 00 5a ]</p>
<p>
Parameter Compatibility Information:</p>
<p>
[ 39 02 2d c0 ]</p>
<p> </p>
<p>Unhandled optional parameter 0x2d 'Unknown'</p>
<p>[0x0 0x5a ]</p>
<p>Unhandled optional parameter 0x39 'Parameter
Compatibility Information'</p>
<p>[0x2d 0xc0 ]</p>
<p> -- DAHDI/63-1 answered
SIP/4735211000-00000036 </p>
<p> </p>
<p> </p>
<p>localhost*CLI> localhost*CLI></p>
<p> </p>
<p> </p>
<p>
#*
( # = Audio Level * = Max Audio Hit )</p>
<p><----------------(RX)---------------->
<----------------(TX)----------------></p>
<p>
###################
*
Rx: 16 ( 16) Tx: 4570 ( 4850)</p>
<p>
########################*
Rx: 16 ( 16) Tx: 1447 ( 1474)</p>
<p>
#*
Rx: 16 ( 16) Tx: 98
( 98)</p>
<p> </p>
<p class="MsoNormal"><span style="font-size:11.0pt;color:#1F497D"> </span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;color:#1F497D">Thanks,</span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;color:#1F497D">Dave</span></p>
<div>
<p class="MsoNormal"> </p>
</div>
</div></div></div>
</div>
<br>--<br>
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