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<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>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.<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Thanks,<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Dave<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<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;font-family:"Tahoma","sans-serif"'>From:</span></b><span
style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'> asterisk-ss7-bounces@lists.digium.com
[mailto:asterisk-ss7-bounces@lists.digium.com] <b>On Behalf Of </b>Abdul Basit<br>
<b>Sent:</b> Tuesday, November 09, 2010 3:41 PM<br>
<b>To:</b> asterisk-ss7@lists.digium.com<br>
<b>Subject:</b> Re: [asterisk-ss7] No Audio on SS7 calls to Remote PRIs<o:p></o:p></span></p>
</div>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal>Please past your chan_dahdi.conf and system.conf.<o:p></o:p></p>
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<p class=MsoNormal><o:p> </o:p></p>
</div>
<div>
<p class=MsoNormal>Also check if you have ulaw selected in your sip phone.<o:p></o:p></p>
</div>
<div>
<p class=MsoNormal><o:p> </o:p></p>
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<p class=MsoNormal style='margin-bottom:12.0pt'><o:p> </o:p></p>
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<p class=MsoNormal>On Wed, Nov 10, 2010 at 12:35 AM, dave george <<a
href="mailto:dgeorge@teletoneinc.com">dgeorge@teletoneinc.com</a>> wrote:<o:p></o:p></p>
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<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><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><o:p></o:p></p>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style='font-size:11.0pt;color:#1F497D'> </span><o:p></o:p></p>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style='font-size:11.0pt;color:#1F497D'>Thanks,</span><o:p></o:p></p>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style='font-size:11.0pt;color:#1F497D'>Dave</span><o:p></o:p></p>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style='font-size:11.0pt;color:#1F497D'> </span><o:p></o:p></p>
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<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><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><o:p></o:p></p>
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</div>
<div>
<div>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <o:p></o:p></p>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style='font-size:11.0pt;color:#1F497D'>Hi Guys,</span><o:p></o:p></p>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style='font-size:11.0pt;color:#1F497D'> </span><o:p></o:p></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><o:p></o:p></p>
<p><span style='font-size:11.0pt;color:#1F497D'>Example </span>Address signals:
5339427#<o:p></o:p></p>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style='font-size:11.0pt;color:#1F497D'> </span><o:p></o:p></p>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style='font-size:11.0pt;color:#1F497D'> </span><o:p></o:p></p>
<p>libss7 version: 1.0.2<o:p></o:p></p>
<p>DAHDI Version: 2.4.0 Echo Canceller: MG2<o:p></o:p></p>
<p> <o:p></o:p></p>
<p><span style='color:#1F497D'> </span><o:p></o:p></p>
<p> Network Indicator: 2 Priority: 0
User Part: ISUP (5)<o:p></o:p></p>
<p> [ 85 ]<o:p></o:p></p>
<p> OPC 2500 DPC 5352 SLS 0<o:p></o:p></p>
<p> [ e8 14 71 02 ]<o:p></o:p></p>
<p>
CIC: 400<o:p></o:p></p>
<p>
[ 90 01 ]<o:p></o:p></p>
<p>
Message Type: CPG<o:p></o:p></p>
<p>
[ 2c ]<o:p></o:p></p>
<p>
--FIXED LENGTH PARMS[1]--<o:p></o:p></p>
<p>
Event
Information:<o:p></o:p></p>
<p>
ALERTING<o:p></o:p></p>
<p>
[ 01 ]<o:p></o:p></p>
<p>
--OPTIONAL PARMS--<o:p></o:p></p>
<p>
Backward Call Indicator:<o:p></o:p></p>
<p>
Charge indicator: 2<o:p></o:p></p>
<p>
Called party's status indicator: 1<o:p></o:p></p>
<p>
Called party's category indicator: 1<o:p></o:p></p>
<p>
End to End method indicator: 0<o:p></o:p></p>
<p>
Interworking indicator: 0<o:p></o:p></p>
<p>
End to End information indicator: 0<o:p></o:p></p>
<p>
ISDN user part indicator: 1<o:p></o:p></p>
<p>
Holding indicator: 0<o:p></o:p></p>
<p>
ISDN access indicator: 1<o:p></o:p></p>
<p>
Echo control device indicator: 1<o:p></o:p></p>
<p>
SCCP method indicator: 0<o:p></o:p></p>
<p>
[
11 02 16 34 ]<o:p></o:p></p>
<p>
Optional Backward Call Indicator:<o:p></o:p></p>
<p>
In-band information indicator: 1<o:p></o:p></p>
<p>
Call diversion may occur indicator: 1<o:p></o:p></p>
<p>
Simple segmentation indicator: 0<o:p></o:p></p>
<p>
MLPP user indicator: 0<o:p></o:p></p>
<p>
[ 29 01 03 ]<o:p></o:p></p>
<p> <o:p></o:p></p>
<p> -- DAHDI/63-1 is ringing<o:p></o:p></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 ]<o:p></o:p></p>
<p>FSN: 28 FIB 1<o:p></o:p></p>
<p>BSN: 58 BIB 1<o:p></o:p></p>
<p><[1] MSU<o:p></o:p></p>
<p>[ ba 9c 16 ]<o:p></o:p></p>
<p> Network Indicator: 2 Priority: 0
User Part: ISUP (5)<o:p></o:p></p>
<p> [ 85 ]<o:p></o:p></p>
<p> OPC 2500 DPC 5352 SLS 0<o:p></o:p></p>
<p> [ e8 14 71 02 ]<o:p></o:p></p>
<p>
CIC: 400<o:p></o:p></p>
<p>
[ 90 01 ]<o:p></o:p></p>
<p>
Message Type: ANM<o:p></o:p></p>
<p>
[ 09 ]<o:p></o:p></p>
<p>
--OPTIONAL PARMS--<o:p></o:p></p>
<p>
Backward Call Indicator:<o:p></o:p></p>
<p>
Charge indicator: 0<o:p></o:p></p>
<p>
Called party's status indicator: 1<o:p></o:p></p>
<p>
Called party's category indicator: 0<o:p></o:p></p>
<p>
End to End method indicator: 0<o:p></o:p></p>
<p>
Interworking indicator: 0<o:p></o:p></p>
<p>
End to End information indicator: 0<o:p></o:p></p>
<p>
ISDN user part indicator: 1<o:p></o:p></p>
<p>
Holding indicator: 0<o:p></o:p></p>
<p>
ISDN
access indicator: 1<o:p></o:p></p>
<p>
Echo control device indicator: 1<o:p></o:p></p>
<p>
SCCP method indicator: 0<o:p></o:p></p>
<p>
[ 11 02 04 34 ]<o:p></o:p></p>
<p>
Unknown Parameter (0x2d):<o:p></o:p></p>
<p>
[ 00 5a ]<o:p></o:p></p>
<p>
Parameter Compatibility Information:<o:p></o:p></p>
<p>
[ 39 02 2d c0 ]<o:p></o:p></p>
<p> <o:p></o:p></p>
<p>Unhandled optional parameter 0x2d 'Unknown'<o:p></o:p></p>
<p>[0x0 0x5a ]<o:p></o:p></p>
<p>Unhandled optional parameter 0x39 'Parameter Compatibility Information'<o:p></o:p></p>
<p>[0x2d 0xc0 ]<o:p></o:p></p>
<p> -- DAHDI/63-1 answered SIP/4735211000-00000036 <o:p></o:p></p>
<p> <o:p></o:p></p>
<p> <o:p></o:p></p>
<p>localhost*CLI> localhost*CLI><o:p></o:p></p>
<p> <o:p></o:p></p>
<p> <o:p></o:p></p>
<p>
#*
( # = Audio Level * = Max Audio Hit )<o:p></o:p></p>
<p><----------------(RX)---------------->
<----------------(TX)----------------><o:p></o:p></p>
<p>
###################
*
Rx: 16 ( 16) Tx: 4570 ( 4850)<o:p></o:p></p>
<p>
########################*
Rx: 16 ( 16) Tx: 1447 ( 1474)<o:p></o:p></p>
<p>
#*
Rx: 16 ( 16) Tx: 98
( 98)<o:p></o:p></p>
<p> <o:p></o:p></p>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style='font-size:11.0pt;color:#1F497D'> </span><o:p></o:p></p>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style='font-size:11.0pt;color:#1F497D'>Thanks,</span><o:p></o:p></p>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style='font-size:11.0pt;color:#1F497D'>Dave</span><o:p></o:p></p>
<div>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <o:p></o:p></p>
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<p class=MsoNormal><br>
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<p class=MsoNormal><br>
<br clear=all>
<br>
-- <o:p></o:p></p>
<div>
<p class=MsoNormal>Regards,<o:p></o:p></p>
</div>
<p class=MsoNormal><br>
Abdul Basit | +92 32 1416 4196<o:p></o:p></p>
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