01-27-2012 09:25 AM - edited 03-16-2019 09:15 AM
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!
Solved! Go to Solution.
01-27-2012 12:02 PM
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
01-27-2012 09:44 AM
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
01-27-2012 11:13 AM
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
01-27-2012 12:02 PM
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
01-27-2012 12:22 PM
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?
01-27-2012 12:27 PM
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
01-27-2012 12:35 PM
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.
01-27-2012 02:02 PM
Chris
Nice work here.
I am trying to brush up on SIP too.
(+5 )
Thanks
Regards
Alex
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