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CME 8.6 Inbound SIP Oddities

dbarsky88
Level 1
Level 1

Hi everyone,

I have a cisco 3825 running 15.1(3) with CME 8.6. I have a sip trunk registered to a voip.ms SIP account and 2 voip.ms DID's that I intend to use for inbound calling to the system. A few things to note right off the bat:

  • Voip.ms SIP accounts are registering fine
  • I can place outbound calls and everything behaves as expected (ie, translation pattern generates outbound CID etc)
  • I have used multiple online docs for CME to Voip.ms config samples and I can't seem to get inbound calls to work.
  • As far as call flow is concerned the goal is to have the call come to an inbound dial peer (dial-peer voice 10 voip) and then be translated to the number of one of my ephone DN's (see translation-rule 8)
  • When calling either of my DID's the result is a fast-busy.

Please see the relevant parts of my config for details:

voice service voip

ip address trusted list

  ipv4 0.0.0.0 0.0.0.0

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

sip

  registrar server expires max 250 min 200

  no call service stop

!

voice class codec 1

codec preference 1 g711ulaw

codec preference 2 g711alaw

!

!

voice register global

max-dn 100

max-pool 100

!

!

!

voice translation-rule 1

rule 1 /\(..........\)/ /9\1/

!

voice translation-rule 4

rule 1 /91+/ /1/

!

voice translation-rule 5

rule 1 /9011+/ /011/

!

voice translation-rule 7

rule 1 /41../ /12222222222/

rule 2 /43../ /12222222222/

rule 3 /44../ /12222244444/

rule 4 /42../ /12222255555/

!

voice translation-rule 8

rule 1 /2222225555/ /1000/

rule 2 /2222224444/ /1001/

!

!

voice translation-profile CALLER-ID

translate calling 7

!

voice translation-profile INCOMING

translate called 8

!

voice translation-profile INTERNATIONAL

translate calling 7

translate called 5

!

voice translation-profile OUT11DIGIT

translate calling 7

translate called 4

!

dial-peer voice 10 voip

description [BDG] VOIP.MS 11 DIGIT OUT

translation-profile incoming INCOMING

translation-profile outgoing OUT11DIGIT

destination-pattern 91[2-9]..[2-9]......

session protocol sipv2

session target dns:chicago.voip.ms

incoming called-number .

dtmf-relay rtp-nte

codec g711ulaw

no vad

!

dial-peer voice 11 voip

description [BDG] VOIP.MS INTERNATIONAL OUT

translation-profile outgoing INTERNATIONAL

destination-pattern 9011T

session protocol sipv2

session target dns:chicago.voip.ms

dtmf-relay rtp-nte

codec g711ulaw

no vad

!

sip-ua

credentials username AAAAAAAA password 7 XXXXXXXXXXXX realm chicago.voip.ms

authentication username AAAAAAAA password 7 XXXXXXXXXXXXX

nat symmetric role active

nat symmetric check-media-src

no remote-party-id

retry invite 3

retry register 3

timers register 150

registrar 1 dns:chicago.voip.ms expires 300

sip-server dns:chicago.voip.ms

connection-reuse

!

!

I ran a debug voip dialpeer and here is the result during an inbound call. What I don't understand is why CME is attemping to match an inbound call to an outgoing dial peer. See below:

Aug 13 17:38:11.592: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

   Calling Number=7736735515, Called Number=7736735515, Peer Info Type=DIALPEER_INFO_SPEECH

Aug 13 17:38:11.592: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

   Match Rule=DP_MATCH_DEST; Called Number=7736735515

Aug 13 17:38:11.592: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

   No Outgoing Dial-peer Is Matched; Result=NO_MATCH(-1)

Aug 13 17:38:11.592: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:

   dialstring=7736735515, saf_enabled=1, saf_dndb_lookup=1, dp_result=-1

Aug 13 17:38:11.592: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersMoreArg:

   Result=NO_MATCH(-1)

Oddly enough, if I add two dial-peers like below (since it seems its looking for an outbound dial peer) the configuration works, but this is messy and a workaround.

dial-peer voice 20 voip

destination-pattern 2222225555

session protocol sipv2

session target ipv4:10.1.20.2

dtmf-relay rtp-nte

no vad

!

dial-peer voice 21 voip

destination-pattern 2222224444

session protocol sipv2

session target ipv4:10.1.20.2

dtmf-relay rtp-nte

no vad

!

Does anyone have any ideas as to why this is happening?

Thanks!

18 Replies 18

paolo bevilacqua
Hall of Fame
Hall of Fame

Because the translation-rule is, somehow wrong. I cannot say why is wrong without seeing the full, unedited configuration,

Hmm, that would make sense... would you mind if I PM you the full unedited config? I've seen your posts on here before and you're very frequently spot-on

Please understand that private support is for my customers only. You can probably understand what is wrong with some observation.

Ok, I understand.

What I don't understand is in the debug output it's showing that the called number is indeed exactly what the translation pattern expects (I realize it looks off in the config I posted above since I obscured the numbers in the translation output but not in the debug output) and it still doesn't route the call the the extension, the result is still a fast-busy.

Not sure, you can take some more debugs or as corner case can be a bug.

dbarsky88
Level 1
Level 1

Yeah, very strange -- I just ran a debug ccsip messages because I wasn't sure whether the DNIS digits sent by Voip.ms matched my rules and if you look at the invite message (below) you'll see that it is hitting the same number as my translation pattern expects.

Aug 13 19:14:06.436: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

INVITE sip:7736735515@OBSCUREDIP:1031 SIP/2.0

Via: SIP/2.0/UDP 64.120.22.242:5060;branch=z9hG4bK0843b058;rport

From: "2245587339" <2245587339>;tag=as75401b90

To: <7736735515>

Contact: <2245587339>

Call-ID: 4210a49116601a957d03f1803a0ada6f@64.120.22.242

CSeq: 102 INVITE

User-Agent: VoIPMS/SERAST

Max-Forwards: 70

Remote-Party-ID: "2245587339" <2245587339>;privacy=off;screen=no

Date: Mon, 13 Aug 2012 19:14:06 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO

Supported: replaces

Content-Type: application/sdp

Content-Length: 318

Please configure

dial-peer voice 10

no translation-profile incoming INCOMING

no incoming called-number

dial-peer voice 1 voip

translation-profile incoming INCOMING

session protocol sipv2

incoming called-number .

dtmf-relay rtp-nte

codec g711ulaw

no vad

Ok,

So I have reconfigured my dial-peers as you show above and have tried placing an inbound call while running another debug voip dialpeer -- same result as before:

Aug 13 19:28:07.018: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

   Calling Number=7736735515, Called Number=7736735515, Peer Info Type=DIALPEER_INFO_SPEECH

Aug 13 19:28:07.018: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

   Match Rule=DP_MATCH_DEST; Called Number=7736735515

Aug 13 19:28:07.018: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

   No Outgoing Dial-peer Is Matched; Result=NO_MATCH(-1)

Aug 13 19:28:07.018: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:

   dialstring=7736735515, saf_enabled=1, saf_dndb_lookup=1, dp_result=-1

Aug 13 19:28:07.018: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersMoreArg:

   Result=NO_MATCH(-1)

But the called number is 7736735515 correct?

why the translation is  rule 1 /2222225555/ /1000/ ?

If the PSTN Caller call this number 7736735515 will not match this translation

Can you try this on your translation

voice translation-rule 8

rule 1 /7736735515/ /1000/

Regards

Leonardo Santana

Regards
Leonardo Santana

*** Rate All Helpful Responses***

leonardotadeu,

My apologies for the translation rule confusion -- I had obscured the number in the config I posted initially and forgot to change it in my debug that I posted too. My rule is actually exactly as you have posted above

Hi Dmitry, if the rules are ok...

I had a similar problem in a sip trunk with service provider.

.

My solution was this command.

voice service voip

sip

  g729 annexb-all

!

voice class codec 1

codec preference 1 g729br8 bytes 30

codec preference 2 g729r8 bytes 30

codec preference 3 g711ulaw

!

!

dial-peer voice 1001 voip

translation-profile incoming INCOMING

session protocol sipv2

incoming called-number .

dtmf-relay rtp-nte

voice-class codec 1

no vad

!

Hope this help!

amendozar,

Thanks for the thorough reply - I added this config to my 3825 (with fingers crossed) and alas, the result was still the same. So far, the only way I've gotten inbound calling to work is with the following config (which in my eyes makes no sense as to how or why it actually works, and it's a bit inconsistent/wrong). It's almost like the call terminates on the CME and attempts to redirect back out and then I have to trap it with an outbound peer and redirect it back inside to dial-peer 10.... this ends up being really sloppy though, and it doesn't make any sense as far as call flow is concerned.

dial-peer voice 11 voip

description [BDG] VOIP.MS INTERNATIONAL OUT

translation-profile outgoing INTERNATIONAL

destination-pattern 9011T

session protocol sipv2

session target dns:chicago.voip.ms

dtmf-relay rtp-nte

codec g711ulaw

no vad

!

dial-peer voice 10 voip

description [BDG] VOIP.MS 11 DIGIT OUT

translation-profile incoming INCOMING

translation-profile outgoing OUT11DIGIT

destination-pattern 91[2-9]..[2-9]......

session protocol sipv2

session target dns:chicago.voip.ms

incoming called-number .

voice-class codec 1

dtmf-relay rtp-nte

no vad

!

dial-peer voice 20 voip

destination-pattern 8476753705

session protocol sipv2

session target ipv4:10.1.20.2

dtmf-relay rtp-nte

no vad

!

dial-peer voice 21 voip

destination-pattern 7736735515

session protocol sipv2

session target ipv4:10.1.20.2

dtmf-relay rtp-nte

no vad

!

The incoming DP matching is failing.

Can you post the complete invite trace, and a more detailed dialpeer debug.

Paolo,

Sure, I can definitely get that for you. Do you recommend any specific debug commands? I've been using debug voip dialpeer and debug ccsip all.

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