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Replies

Incomming calls from internal voip dialpeer with sip trunk do not ring FXS line

Zachary Harris
Level 1
Level 1

I am sure i am missing something really simple. I have a voice lab setup where it would be cheaper to use my old cme 2600xm router voip dial peer for the FXS lines and my newer dial peer 2811 for the IP phones and SIP trunking.

Outbound calls to the sip trunk from the IP phones on the 2811 router work fine. Outbound calls to the sip trunk from the 2621xm router with the

FXS lines work fine.

Incoming calls from the sip trunk when translated to the line of any IP phone registered to the 2811 router work fine.

The problem is that when I change the translation from :

rule 1 /708xxxxxxx/ /1001/ (line on IP phone on cme 2811 router)

to

rule 1 /708xxxxxxx/ /1011/  (FXS line on cme 2621xm router)

and try an incoming call from the sip trunk, I get no ring and after 20+ seconds, I get a busy signal.

A debug from the 2621 router shows that the 2811 router is routing the call correctly (I think):

Jan 27 17:15:23.341: //-1/54F1F0798396/DPM/dpAssociateIncomingPeerCore:

   Calling Number=773xxxxxxx, Called Number=1011, Voice-Interface=0x0,

   Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,

   Peer Info Type=DIALPEER_INFO_SPEECH

Jan 27 17:15:23.341: //-1/54F1F0798396/DPM/dpAssociateIncomingPeerCore:

   Result=NO_MATCH(-1) After All Match Rules Attempt

Jan 27 17:15:23.349: //-1/54F1F0798396/DPM/dpAssociateIncomingPeerCore:

   Calling Number=773xxxxxxx, Called Number=1011, Voice-Interface=0x0,

   Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,

   Peer Info Type=DIALPEER_INFO_SPEECH

Jan 27 17:15:23.349: //-1/54F1F0798396/DPM/dpAssociateIncomingPeerCore:

   Result=NO_MATCH(-1) After All Match Rules Attempt

Jan 27 17:15:23.365: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

   Calling Number=1011, Called Number=1011, Peer Info Type=DIALPEER_INFO_SPEECH

Jan 27 17:15:23.365: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

   Match Rule=DP_MATCH_DEST; Called Number=1011

Jan 27 17:15:23.369: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

   Result=Success(0) after DP_MATCH_DEST

Jan 27 17:15:23.369: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersMoreArg:

   Result=SUCCESS(0)

   List of Matched Outgoing Dial-peer(s):

     1: Dial-peer Tag=3001

Jan 27 17:15:23.369: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:

   Calling Number=1011, Called Number=, Voice-Interface=0x0,

   Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,

   Peer Info Type=DIALPEER_INFO_SPEECH

Jan 27 17:15:23.369: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:

   Result=NO_MATCH(-1) After All Match Rules Attempt

Jan 27 17:15:23.377: //-1/54F1F0798396/DPM/dpMatchPeersCore:

   Calling Number=, Called Number=1011, Peer Info Type=DIALPEER_INFO_SPEECH

Jan 27 17:15:23.377: //-1/54F1F0798396/DPM/dpMatchPeersCore:

   Match Rule=DP_MATCH_DEST; Called Number=1011

CME_Router#un all

Jan 27 17:15:23.377: //-1/54F1F0798396/DPM/dpMatchPeersCore:

   Result=Success(0) after DP_MATCH_DEST

Jan 27 17:15:23.381: //-1/54F1F0798396/DPM/dpMatchPeersMoreArg:

   Result=SUCCESS(0)

   List of Matched Outgoing Dial-peer(s):

     1: Dial-peer Tag=3001

Here are the voice configs for both routers:

2621xm:

interface FastEthernet0/0

ip address 192.168.1.123 255.255.255.0

duplex auto

speed auto

dial-peer voice 3001 pots

destination-pattern 1011

port 1/1/0

!

dial-peer voice 3002 pots

destination-pattern 1012

port 1/1/1

!

dial-peer voice 10 voip

destination-pattern 100.

session target ipv4:192.168.1.125

dtmf-relay cisco-rtp

codec g711ulaw

no vad

!

dial-peer voice 999 voip

destination-pattern 91..........

session target ipv4:192.168.1.125

dtmf-relay cisco-rtp

codec g711ulaw

no vad

2811:

voice translation-rule 3

rule 1 /708xxxxxxxx/ /1011/

voice translation-profile 31

translate called 3

interface FastEthernet0/0

ip address 192.168.1.125 255.255.255.0

ip pim sparse-dense-mode

duplex auto

speed auto

dial-peer voice 300 voip

destination-pattern 101.

session target ipv4:192.168.1.123

dtmf-relay cisco-rtp

codec g711ulaw

no vad

!

dial-peer voice 31 voip

translation-profile incoming 31

incoming called-number 708.......

dtmf-relay rtp-nte

!

I have been trying since Wed to find similar questions on google and here but havnt been able to. PLEASE HELP. Thanks!

1 Accepted Solution

Accepted Solutions

Can you force the code to be g711 all the way, you need one of the

26XX

dial-peer voice 10 voip

incoming called-number .

codec g711ulaw

2811:

dial-peer voice 31 voip

codec g711ulaw

If that does not do anything can you post "debug ccsip messages" from the 2811?

Chris

View solution in original post

7 Replies 7

Chris Deren
Hall of Fame
Hall of Fame

What happens when you call from IP phone to the FXS extension?

Since you are doing SIP to H323 transfers do you have the followin on both routers:

voice service voip

allow-connections h323 to h323

allow-connections h323 to sip

allow-connections sip to h323

Chris

Thanks for replying!

Calls from IP to FXS work fine and calls from FXS to IP phone work fine as well.

for the 2621xm:

voice service voip

allow-connections h323 to h323

allow-connections h323 to sip

allow-connections sip to h323

allow-connections sip to sip

supplementary-service h450.12

for the 2811:

voice service voip

gcid

clid substitute name

allow-connections h323 to h323

allow-connections h323 to sip

allow-connections sip to h323

allow-connections sip to sip

supplementary-service h450.12

no supplementary-service sip moved-temporarily

no supplementary-service sip refer

fax protocol cisco

sip

  e911

  registrar server expires max 250 min 200

  transport switch udp tcp

  asserted-id ppi

Can you force the code to be g711 all the way, you need one of the

26XX

dial-peer voice 10 voip

incoming called-number .

codec g711ulaw

2811:

dial-peer voice 31 voip

codec g711ulaw

If that does not do anything can you post "debug ccsip messages" from the 2811?

Chris

Chris you are the man!

Hard setting the codec on the 2811 to g711ulaw did the trick! It rang on thru to the FXS lines!

2811:

dial-peer voice 31 voip

translation-profile incoming 31

incoming called-number 708xxxxxxxx

dtmf-relay rtp-nte

codec g711ulaw

I had been fighting with this for 2 days and one little command fixed it. Again many thanks.

So i guess now the question is, doesnt the codec auto negotiate on the SIP trunk?

No, when you get SIP trunk from carrier you decide on which codec you will use, typically G711 or G729, if you need to change that internally you will require transcoders.

Glad, the change fixed it for you, too bad you did not find it that helpful since you rated it low.

Chris

Gotcha. Very good to know.

Sorry!!! I was trying to click the box to type and hit that by accident. Can it be changed? First time posting.

Chris

Nice work here.

I am trying to brush up on SIP too.

(+5 )

Thanks

Regards

Alex

Regards, Alex. Please rate useful posts.
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