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Hi,<br>
<br>
configuration of R2 and E1 port are working now. And I also
configured dial-peer for incoming and outgoing see it below<br>
<br>
controller E1 2/2<br>
framing NO-CRC4<br>
ds0-group 1 timeslots 1-15,17-31 type r2-digital r2-compelled ani<br>
<br>
<br>
dial-peer voice 2 pots<br>
incoming called-number .<br>
no digit-strip<br>
direct-inward-dial<br>
port 2/2:1<br>
!<br>
dial-peer voice 1626 voip<br>
description to SYSmaster<br>
destination-pattern 1626<br>
voice-class codec 2<br>
session target ipv4:x.x.x.x IP address modified<br>
<br>
but when I dialed to access number 1626 then I got log from debug<br>
<br>
debug voip dialpeer inout is ON (filter is OFF)<br>
<br>
<blockquote type="cite">.Nov 15 16:27:16.484 GMT:
//-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:<br>
Calling Number=11319722, Called Number=1626,
Voice-Interface=0x69E1D254,<br>
Timeout=TRUE, Peer Encap Type=ENCAP_VOICE, Peer Search
Type=PEER_TYPE_VOICE,<br>
Peer Info Type=DIALPEER_INFO_SPEECH<br>
.Nov 15 16:27:16.484 GMT:
//-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:<br>
Result=Success(0) after DP_MATCH_INCOMING_DNIS; Incoming
Dial-peer=2<br>
.Nov 15 16:27:16.484 GMT:
//-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:<br>
Calling Number=11319722, Called Number=1626,
Voice-Interface=0x0,<br>
Timeout=TRUE, Peer Encap Type=ENCAP_VOICE, Peer Search
Type=PEER_TYPE_VOICE,<br>
Peer Info Type=DIALPEER_INFO_SPEECH<br>
.Nov 15 16:27:16.484 GMT:
//-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:<br>
Result=Success(0) after DP_MATCH_INCOMING_DNIS; Incoming
Dial-peer=2<br>
.Nov 15 16:27:16.484 GMT:
//-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:<br>
Calling Number=11319722, Called Number=1626,
Voice-Interface=0x69E1D254,<br>
Timeout=TRUE, Peer Encap Type=ENCAP_VOICE, Peer Search
Type=PEER_TYPE_VOICE,<br>
Peer Info Type=DIALPEER_INFO_SPEECH<br>
.Nov 15 16:27:16.484 GMT:
//-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:<br>
Result=Success(0) after DP_MATCH_INCOMING_DNIS; Incoming
Dial-peer=2<br>
.Nov 15 16:27:16.548 GMT: //-1/76025A0C99C2/DPM/dpMatchPeersCore:<br>
Calling Number=, Called Number=1626, Peer Info
Type=DIALPEER_INFO_SPEECH<br>
.Nov 15 16:27:16.548 GMT: //-1/76025A0C99C2/DPM/dpMatchPeersCore:<br>
Match Rule=DP_MATCH_DEST; Called Number=1626<br>
.Nov 15 16:27:16.548 GMT: //-1/76025A0C99C2/DPM/dpMatchPeersCore:<br>
Result=Success(0) after DP_MATCH_DEST<br>
.Nov 15 16:27:16.548 GMT:
//-1/76025A0C99C2/DPM/dpMatchPeersMoreArg:<br>
Result=SUCCESS(0)<br>
List of Matched Outgoing Dial-peer(s):<br>
1: Dial-peer Tag=1626<br>
.Nov 15 16:27:16.548 GMT: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:<br>
Calling Number=1626, Called Number=1626, Peer Info
Type=DIALPEER_INFO_SPEECH<br>
.Nov 15 16:27:16.548 GMT: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:<br>
Match Rule=DP_MATCH_DEST; Called Number=1626<br>
.Nov 15 16:27:16.548 GMT: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:<br>
Result=Success(0) after DP_MATCH_DEST<br>
.Nov 15 16:27:16.548 GMT:
//-1/xxxxxxxxxxxx/DPM/dpMatchPeersMoreArg:<br>
Result=SUCCESS(0)<br>
List of Matched Outgoing Dial-peer(s):<br>
1: Dial-peer Tag=1626<br>
.Nov 15 16:27:16.548 GMT: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:<br>
Calling Number=1626, Called Number=1626, Peer Info
Type=DIALPEER_INFO_SPEECH<br>
.Nov 15 16:27:16.548 GMT: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:<br>
Match Rule=DP_MATCH_DEST; Called Number=1626<br>
.Nov 15 16:27:16.548 GMT: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:<br>
Result=Success(0) after DP_MATCH_DEST<br>
.Nov 15 16:27:16.548 GMT:
//-1/xxxxxxxxxxxx/DPM/dpMatchPeersMoreArg:<br>
Result=SUCCESS(0)<br>
List of Matched Outgoing Dial-peer(s):<br>
1: Dial-peer Tag=1626<br>
.Nov 15 16:27:16.548 GMT: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:<br>
Calling Number=, Called Number=1626, Peer Info
Type=DIALPEER_INFO_SPEECH<br>
.Nov 15 16:27:16.548 GMT: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:<br>
Match Rule=DP_MATCH_DEST; Called Number=1626<br>
.Nov 15 16:27:16.548 GMT: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:<br>
Result=Success(0) after DP_MATCH_DEST<br>
.Nov 15 16:27:16.548 GMT:
//-1/xxxxxxxxxxxx/DPM/dpMatchPeersMoreArg:<br>
Result=SUCCESS(0)<br>
List of Matched Outgoing Dial-peer(s):<br>
1: Dial-peer Tag=1626<br>
.Nov 15 16:27:16.548 GMT: //-1/76025A0C99C2/DPM/dpMatchPeersCore:<br>
Calling Number=, Called Number=1626, Peer Info
Type=DIALPEER_INFO_SPEECH<br>
.Nov 15 16:27:16.548 GMT: //-1/76025A0C99C2/DPM/dpMatchPeersCore:<br>
Match Rule=DP_MATCH_DEST; Called Number=1626<br>
.Nov 15 16:27:16.548 GMT: //-1/76025A0C99C2/DPM/dpMatchPeersCore:<br>
Result=Success(0) after DP_MATCH_DEST<br>
.Nov 15 16:27:16.548 GMT:
//-1/76025A0C99C2/DPM/dpMatchPeersMoreArg:<br>
Result=SUCCESS(0)<br>
List of Matched Outgoing Dial-peer(s):<br>
1: Dial-peer Tag=1626<br>
.Nov 15 16:27:16.932 GMT:
//-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:<br>
Calling Number=11319722, Called Number=1626,
Voice-Interface=0x69E1D254,<br>
Timeout=TRUE, Peer Encap Type=ENCAP_VOICE, Peer Search
Type=PEER_TYPE_VOICE,<br>
Peer Info Type=DIALPEER_INFO_SPEECH<br>
.Nov 15 16:27:16.932 GMT:
//-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:<br>
Result=Success(0) after DP_MATCH_INCOMING_DNIS; Incoming
Dial-peer=2<br>
</blockquote>
<br>
but nothing on my sysmaster gateway.<br>
<br>
Question : <br>
<br>
Do I hear IVR from sysmaster gateway when I set this configuration
on cisco gateway ?<br>
<br>
Regards,<br>
Tseveen<br>
<br>
On 11/15/2011 15:44, Abebe Amare wrote:
<blockquote
cite="mid:CANrv87iAGOGp2ojhAmkoxKy9+WoBjuSmpQ-eCALLRYZgN9DAzw@mail.gmail.com"
type="cite"><font size="2"><span class="content">
<pre>Hi,
To configure the E1 port for R2 signalling,
AS5350(config)# <b class="cCN_CmdName">controller e1 0/0</b>
AS5350(config-controller)# <b class="cCN_CmdName">ds0-group 1 timeslots 1-30 type r2-analog r2-compelled ani</b><span class="content"><span class="content"></span></span>
the R2 signalling type depends on your PSTN connection.
</pre>
Configure the dial-peers as below:<br>
</span></font>
<div class="pEx1_Example1">
<pre>dial-peer voice 10 pots
incoming called-number .
direct-inward-dial
port 0/0:D
dial-peer voice 2 voip
</pre>
</div>
<font size="2"><a moz-do-not-send="true" name="wp1079070"></a></font>
<div class="pEx1_Example1">
<pre><font size="2"> destination-pattern x... <- this is the access number
</font></pre>
</div>
<font size="2"><a moz-do-not-send="true" name="wp1079071"></a></font>
<p class="pB1_Body1"><font size="2">
session target ipv4:x.x.x.x <- this is the IP of the
SysMaster GW</font></p>
<p class="pB1_Body1"><font size="2">Check the following documents
for more detail<br>
</font></p>
<p class="pB1_Body1"><font size="2"><a moz-do-not-send="true"
href="http://www.cisco.com/en/US/docs/routers/access/as5350/software/configuration/guide/54bas3.html#wpxref17473">http://www.cisco.com/en/US/docs/routers/access/as5350/software/configuration/guide/54bas3.html#wpxref17473</a><br>
</font></p>
<p class="pB1_Body1"><font size="2"><a moz-do-not-send="true"
href="http://www.cisco.com/en/US/docs/routers/access/as5350/software/configuration/guide/54voice_ps501_TSD_Products_Configuration_Guide_Chapter.html">http://www.cisco.com/en/US/docs/routers/access/as5350/software/configuration/guide/54voice_ps501_TSD_Products_Configuration_Guide_Chapter.html</a><br>
</font></p>
<font size="2"><br>
regards,<br>
</font><br>
<div class="gmail_quote">2011/11/15 Цэвээндорж <span dir="ltr"><<a
moz-do-not-send="true" href="mailto:tseveendorj@gmail.com">tseveendorj@gmail.com</a>></span><br>
<blockquote class="gmail_quote" style="margin:0 0 0
.8ex;border-left:1px #ccc solid;padding-left:1ex;">
Hello,<br>
<br>
I don't know how called it technically. Let me try to explain
what I'm trying to do.<br>
I have SysMaster SM7000 gateway and billing also. I cannot
connect to PSTN via E1 with R2 signaling. I thought I cannot
configure sysmaster correctly then I ask from sysmaster guys
but reply as follow<br>
<br>
<blockquote class="gmail_quote" style="margin:0 0 0
.8ex;border-left:1px #ccc solid;padding-left:1ex">
Basically, we can support this requirement you have but NOT
on the current HW you have.<br>
<br>
If you need to support MFC/R2 signaling with your current
SM7000, then it will cost you just as much for custom
development as it would to just purchase a new SM7000 GW
that can support it already.<br>
</blockquote>
<br>
Now let's begin<br>
I have cisco gateway AS5350XM (Version 12.4(20)T1) with 4xE1
card using current system. I'm planning interconnect to PSTN
with cisco gateway +1xE1 with R2 signaling. My topology looks
like this.<br>
<br>
Access number ---> PSTN ----------E1--------> Cisco
gateway without IVR ----------TCP/IP---------> Sysmaster
gateway with IVR<br>
<br>
I don't know how to configure when user dials to Access number
(example 1626) then Cisco gateway without IVR receive request
and immediately send to Sysmaster gateway with IVR and
customer should hear the IVR.<br>
<br>
<br>
Any help will be appreciated.<br>
<br>
Regards,<br>
Tseveen.<br>
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</blockquote>
</div>
<br>
</blockquote>
<br>
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