[cisco-voip] dial-peer
Abebe Amare
abucho at gmail.com
Tue Nov 15 04:28:38 EST 2011
Hi,
Add these config to the voip dial-peer
dtmf-relay h245-alphanumeric
no vad
regards,
Abebe
2011/11/15 Цэвээндорж <tseveendorj at gmail.com>
> Hi,
>
> configuration of R2 and E1 port are working now. And I also configured
> dial-peer for incoming and outgoing see it below
>
> controller E1 2/2
> framing NO-CRC4
> ds0-group 1 timeslots 1-15,17-31 type r2-digital r2-compelled ani
>
>
> dial-peer voice 2 pots
> incoming called-number .
> no digit-strip
> direct-inward-dial
> port 2/2:1
> !
> dial-peer voice 1626 voip
> description to SYSmaster
> destination-pattern 1626
> voice-class codec 2
> session target ipv4:x.x.x.x IP address modified
>
> but when I dialed to access number 1626 then I got log from debug
>
> debug voip dialpeer inout is ON (filter is OFF)
>
> .Nov 15 16:27:16.484 GMT:
> //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
> Calling Number=11319722, Called Number=1626, Voice-Interface=0x69E1D254,
> Timeout=TRUE, Peer Encap Type=ENCAP_VOICE, Peer Search
> Type=PEER_TYPE_VOICE,
> Peer Info Type=DIALPEER_INFO_SPEECH
> .Nov 15 16:27:16.484 GMT:
> //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
> Result=Success(0) after DP_MATCH_INCOMING_DNIS; Incoming Dial-peer=2
> .Nov 15 16:27:16.484 GMT:
> //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
> Calling Number=11319722, Called Number=1626, Voice-Interface=0x0,
> Timeout=TRUE, Peer Encap Type=ENCAP_VOICE, Peer Search
> Type=PEER_TYPE_VOICE,
> Peer Info Type=DIALPEER_INFO_SPEECH
> .Nov 15 16:27:16.484 GMT:
> //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
> Result=Success(0) after DP_MATCH_INCOMING_DNIS; Incoming Dial-peer=2
> .Nov 15 16:27:16.484 GMT:
> //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
> Calling Number=11319722, Called Number=1626, Voice-Interface=0x69E1D254,
> Timeout=TRUE, Peer Encap Type=ENCAP_VOICE, Peer Search
> Type=PEER_TYPE_VOICE,
> Peer Info Type=DIALPEER_INFO_SPEECH
> .Nov 15 16:27:16.484 GMT:
> //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
> Result=Success(0) after DP_MATCH_INCOMING_DNIS; Incoming Dial-peer=2
> .Nov 15 16:27:16.548 GMT: //-1/76025A0C99C2/DPM/dpMatchPeersCore:
> Calling Number=, Called Number=1626, Peer Info Type=DIALPEER_INFO_SPEECH
> .Nov 15 16:27:16.548 GMT: //-1/76025A0C99C2/DPM/dpMatchPeersCore:
> Match Rule=DP_MATCH_DEST; Called Number=1626
> .Nov 15 16:27:16.548 GMT: //-1/76025A0C99C2/DPM/dpMatchPeersCore:
> Result=Success(0) after DP_MATCH_DEST
> .Nov 15 16:27:16.548 GMT: //-1/76025A0C99C2/DPM/dpMatchPeersMoreArg:
> Result=SUCCESS(0)
> List of Matched Outgoing Dial-peer(s):
> 1: Dial-peer Tag=1626
> .Nov 15 16:27:16.548 GMT: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
> Calling Number=1626, Called Number=1626, Peer Info
> Type=DIALPEER_INFO_SPEECH
> .Nov 15 16:27:16.548 GMT: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
> Match Rule=DP_MATCH_DEST; Called Number=1626
> .Nov 15 16:27:16.548 GMT: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
> Result=Success(0) after DP_MATCH_DEST
> .Nov 15 16:27:16.548 GMT: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersMoreArg:
> Result=SUCCESS(0)
> List of Matched Outgoing Dial-peer(s):
> 1: Dial-peer Tag=1626
> .Nov 15 16:27:16.548 GMT: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
> Calling Number=1626, Called Number=1626, Peer Info
> Type=DIALPEER_INFO_SPEECH
> .Nov 15 16:27:16.548 GMT: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
> Match Rule=DP_MATCH_DEST; Called Number=1626
> .Nov 15 16:27:16.548 GMT: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
> Result=Success(0) after DP_MATCH_DEST
> .Nov 15 16:27:16.548 GMT: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersMoreArg:
> Result=SUCCESS(0)
> List of Matched Outgoing Dial-peer(s):
> 1: Dial-peer Tag=1626
> .Nov 15 16:27:16.548 GMT: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
> Calling Number=, Called Number=1626, Peer Info Type=DIALPEER_INFO_SPEECH
> .Nov 15 16:27:16.548 GMT: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
> Match Rule=DP_MATCH_DEST; Called Number=1626
> .Nov 15 16:27:16.548 GMT: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
> Result=Success(0) after DP_MATCH_DEST
> .Nov 15 16:27:16.548 GMT: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersMoreArg:
> Result=SUCCESS(0)
> List of Matched Outgoing Dial-peer(s):
> 1: Dial-peer Tag=1626
> .Nov 15 16:27:16.548 GMT: //-1/76025A0C99C2/DPM/dpMatchPeersCore:
> Calling Number=, Called Number=1626, Peer Info Type=DIALPEER_INFO_SPEECH
> .Nov 15 16:27:16.548 GMT: //-1/76025A0C99C2/DPM/dpMatchPeersCore:
> Match Rule=DP_MATCH_DEST; Called Number=1626
> .Nov 15 16:27:16.548 GMT: //-1/76025A0C99C2/DPM/dpMatchPeersCore:
> Result=Success(0) after DP_MATCH_DEST
> .Nov 15 16:27:16.548 GMT: //-1/76025A0C99C2/DPM/dpMatchPeersMoreArg:
> Result=SUCCESS(0)
> List of Matched Outgoing Dial-peer(s):
> 1: Dial-peer Tag=1626
> .Nov 15 16:27:16.932 GMT:
> //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
> Calling Number=11319722, Called Number=1626, Voice-Interface=0x69E1D254,
> Timeout=TRUE, Peer Encap Type=ENCAP_VOICE, Peer Search
> Type=PEER_TYPE_VOICE,
> Peer Info Type=DIALPEER_INFO_SPEECH
> .Nov 15 16:27:16.932 GMT:
> //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
> Result=Success(0) after DP_MATCH_INCOMING_DNIS; Incoming Dial-peer=2
>
>
> but nothing on my sysmaster gateway.
>
> Question :
>
> Do I hear IVR from sysmaster gateway when I set this configuration on
> cisco gateway ?
>
> Regards,
> Tseveen
>
>
> On 11/15/2011 15:44, Abebe Amare wrote:
>
> Hi,
>
> To configure the E1 port for R2 signalling,
>
> AS5350(config)# *controller e1 0/0*
> AS5350(config-controller)# *ds0-group 1 timeslots 1-30 type r2-analog r2-compelled ani*
>
>
>
> the R2 signalling type depends on your PSTN connection.
>
> Configure the dial-peers as below:
>
> dial-peer voice 10 pots
> incoming called-number .
> direct-inward-dial
>
>
> port 0/0:D
>
> dial-peer voice 2 voip
>
> destination-pattern x... <- this is the access number
>
> session target ipv4:x.x.x.x <- this is the IP of the SysMaster GW
>
> Check the following documents for more detail
>
>
> 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/54voice_ps501_TSD_Products_Configuration_Guide_Chapter.html
>
> regards,
>
> 2011/11/15 Цэвээндорж <tseveendorj at gmail.com>
>
>> Hello,
>>
>> I don't know how called it technically. Let me try to explain what I'm
>> trying to do.
>> 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
>>
>> Basically, we can support this requirement you have but NOT on the
>>> current HW you have.
>>>
>>> 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.
>>>
>>
>> Now let's begin
>> 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.
>>
>> Access number ---> PSTN ----------E1--------> Cisco gateway without IVR
>> ----------TCP/IP---------> Sysmaster gateway with IVR
>>
>> 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.
>>
>>
>> Any help will be appreciated.
>>
>> Regards,
>> Tseveen.
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip at puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>
>
>
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