<font size="2"><font face="verdana,sans-serif">Hi Robert,</font></font><div><font size="2"><font face="verdana,sans-serif"><br></font></font></div><div><font size="2"><font face="verdana,sans-serif">Do you need a transcoder? Looking at the output below it looks you do.</font></font></div>
<div><font size="2"><font face="verdana,sans-serif"><br></font></font></div><div><font size="2"><font face="verdana,sans-serif">I would also change the config to this, if transcoding is required you should hear silence as the prompt will play but as the prompt is g711 you should hear nothing. Once you get it sorted and transcoding configured remove the g729 preference.</font></font></div>
<div><font size="2"><font face="verdana,sans-serif"><br></font></font></div><div><font size="2"><font face="verdana,sans-serif"><p class="MsoListParagraphCxSpFirst" style="margin-bottom:0cm;margin-bottom:.0001pt;
mso-add-space:auto"><i style="mso-bidi-font-style:normal"><span style="font-size:8.0pt;line-height:115%;font-family:Consolas;mso-bidi-font-family:
Arial">voice class codec 1</span></i></p>
<p class="MsoListParagraphCxSpMiddle" style="margin-bottom:0cm;margin-bottom:
.0001pt;mso-add-space:auto"><i style="mso-bidi-font-style:normal"><span style="font-size:8.0pt;line-height:115%;font-family:Consolas;mso-bidi-font-family:
Arial"><span style="mso-spacerun:yes"> </span>codec preference 1 g711alaw</span></i></p>
<p class="MsoListParagraphCxSpLast" style="margin-bottom:0cm;margin-bottom:.0001pt;
mso-add-space:auto"><i style="mso-bidi-font-style:normal"><span style="font-size:8.0pt;line-height:115%;font-family:Consolas;mso-bidi-font-family:
Arial"><span style="mso-spacerun:yes"> </span>codec preference 2 g711ulaw</span></i></p><p class="MsoListParagraphCxSpLast" style="margin-bottom:0cm;margin-bottom:.0001pt;
mso-add-space:auto"><i style="mso-bidi-font-style:normal"><span style="font-size:8.0pt;line-height:115%;font-family:Consolas;mso-bidi-font-family:
Arial"> codec preference 3 g729r8</span></i></p><p class="MsoListParagraphCxSpLast" style="margin-bottom:0cm;margin-bottom:.0001pt;
mso-add-space:auto"><i style="mso-bidi-font-style:normal"><span style="font-size:8.0pt;line-height:115%;font-family:Consolas;mso-bidi-font-family:
Arial"><br></span></i></p></font></font></div><div><font face="Consolas"><p class="MsoListParagraphCxSpLast" style="margin-bottom: 0.0001pt; font-size: 11px; line-height: 12px;"><i>Dial-peer voice 1 voip</i></p><p class="MsoListParagraphCxSpLast" style="margin-bottom: 0.0001pt; font-size: 11px; line-height: 12px;">
<i>voice-class codec 20</i></p></font></div><div><font><p class="MsoListParagraphCxSpLast" style="margin-bottom: 0.0001pt; font-family: verdana, sans-serif; font-size: small; "><i style="mso-bidi-font-style:normal"><span style="font-size:8.0pt;line-height:115%;font-family:Consolas;mso-bidi-font-family:
Arial"><br></span></i></p></font></div><div><font size="2"><font face="verdana,sans-serif"><br></font></font></div><div><font size="2"><font face="verdana,sans-serif">cheers,</font></font></div><div><font size="2"><font face="verdana,sans-serif"><br>
</font></font></div><div><font size="2"><font face="verdana,sans-serif">Dan</font></font></div><div><font size="2"><font face="verdana,sans-serif"><br></font></font></div><div><font size="2"><font face="verdana,sans-serif"><br>
</font></font><br><div class="gmail_quote">On Fri, Mar 4, 2011 at 11:05 PM, Robert Hass <span dir="ltr"><<a href="mailto:robhass@gmail.com">robhass@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
Hi<br>
I have problem with dial-peer configuration where incoming calls (VoIP<br>
H.323 and POTS) to specified number should be redirected to TCL<br>
script.<br>
<br>
My configuration:<br>
<br>
voice service voip<br>
allow-connections h323 to h323<br>
fax protocol t38 ls-redundancy 2 hs-redundancy 2 fallback none<br>
h323<br>
sip<br>
call service stop<br>
!<br>
application<br>
service announcement flash:announcement.tcl<br>
!<br>
dial-peer voice 201 pots<br>
service announcement<br>
incoming called-number 911234491<br>
direct-inward-dial<br>
port 0/0/0:15<br>
!<br>
dial-peer voice 300 voip<br>
service announcement<br>
incoming called-number 497<br>
dtmf-relay rtp-nte h245-signal h245-alphanumeric<br>
codec g711ulaw<br>
no vad<br>
!<br>
<br>
And announcement works when incoming call is done from PSTN via ISDN<br>
E1/PRI (Serial0/0/0:15 to number 9112434491) and not works when<br>
incoming call is going via VoIP (H.323 to number 497). I syslog got<br>
cause: DisconnectCause 1 , DisconnectText unassigned number (1)<br>
<br>
What is misconfigured inside 'dial-peer voice 300 voip' ?<br>
<br>
Please help :)<br>
<br>
Robert<br>
<br>
PS. Below is log from ''show call history voice last 1"<br>
<br>
GENERIC:<br>
SetupTime=1966110610 ms<br>
Index=202<br>
PeerAddress=918674488<br>
PeerSubAddress=<br>
PeerId=0<br>
PeerIfIndex=0<br>
LogicalIfIndex=0<br>
DisconnectCause=1<br>
DisconnectText=unassigned number (1)<br>
ConnectTime=0 ms<br>
DisconnectTime=1966110630 ms<br>
CallDuration=00:00:00 sec<br>
CallOrigin=2<br>
ReleaseSource=7<br>
InternalErrorCode=1.1.128.11.5.0<br>
ChargedUnits=0<br>
InfoType=speech<br>
TransmitPackets=0<br>
TransmitBytes=0<br>
ReceivePackets=0<br>
ReceiveBytes=0<br>
VOIP:<br>
ConnectionId[0x58B066A5 0x459611E0 0x93FBAA4D 0x77C09A08]<br>
IncomingConnectionId[0x58B066A5 0x459611E0 0x93FBAA4D 0x77C09A08]<br>
CallID=202<br>
RemoteIPAddress=192.168.100.1<br>
RemoteUDPPort=0<br>
RemoteSignallingIPAddress=192.168.100.1<br>
RemoteSignallingPort=15266<br>
RemoteMediaIPAddress=0.0.0.0<br>
RemoteMediaPort=0<br>
SRTP = off<br>
TextRelay = off<br>
Fallback Icpif=0<br>
Fallback Loss=0<br>
Fallback Delay=0<br>
RoundTripDelay=0 ms<br>
SelectedQoS=best-effort<br>
tx_DtmfRelay=inband-voice<br>
FastConnect=FALSE<br>
<br>
AnnexE=FALSE<br>
<br>
Separate H245 Connection=FALSE<br>
<br>
H245 Tunneling=FALSE<br>
<br>
SessionProtocol=cisco<br>
ProtocolCallId=<br>
SessionTarget=<br>
OnTimeRvPlayout=0<br>
GapFillWithSilence=0 ms<br>
GapFillWithPrediction=0 ms<br>
GapFillWithInterpolation=0 ms<br>
GapFillWithRedundancy=0 ms<br>
HiWaterPlayoutDelay=0 ms<br>
LoWaterPlayoutDelay=0 ms<br>
PlayoutMode = undefined<br>
PlayoutInitialDelay=0 ms<br>
ReceiveDelay=0 ms<br>
LostPackets=0<br>
EarlyPackets=0<br>
LatePackets=0<br>
VAD = enabled<br>
CoderTypeRate=g729r8 pre-ietf<br>
CodecBytes=0<br>
cvVoIPCallHistoryIcpif=0<br>
MediaSetting=flow-around<br>
CallerName=RobertH<br>
CallerIDBlocked=False<br>
OriginalCallingNumber=918674488<br>
OriginalCallingOctet=0x80<br>
OriginalCalledNumber=497<br>
OriginalCalledOctet=0x80<br>
OriginalRedirectCalledNumber=<br>
OriginalRedirectCalledOctet=0xFF<br>
TranslatedCallingNumber=918674488<br>
TranslatedCallingOctet=0x80<br>
TranslatedCalledNumber=497<br>
TranslatedCalledOctet=0x80<br>
TranslatedRedirectCalledNumber=<br>
TranslatedRedirectCalledOctet=0xFF<br>
GwReceivedCalledNumber=497<br>
GwReceivedCalledOctet3=0x80<br>
GwReceivedCallingNumber=918674488<br>
GwReceivedCallingOctet3=0x80<br>
GwReceivedCallingOctet3a=0x0<br>
MediaInactiveDetected=no<br>
MediaInactiveTimestamp=<br>
MediaControlReceived=<br>
LongDurationCallDetected=no<br>
LongDurationCallTimerStamp=<br>
LongDurationCallDuration=<br>
Username=<br>
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</blockquote></div><br></div>