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rmv72
Beginner

CME+SIP TRUNK

Hi,

I can not understand where is problem

i've CME and SIP-trunk to SP. Incoming calls from SP don't received by ip-phone registered on CME.

Here config-

voice service voip

allow-connections h323 to h323

allow-connections h323 to sip

allow-connections sip to h323

allow-connections sip to sip

fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none

sip     

  bind control source-interface GigabitEthernet0/0

  bind media source-interface GigabitEthernet0/0

!

voice class codec 1

codec preference 1 g711ulaw

!        

!

!        

!

voice translation-rule 1

rule 1 /310817/ /1412/

rule 2 /310816/ /1412/

rule 3 /310814/ /1412/

!        

!

voice translation-profile FROM_SIP

translate called 1

dial-peer voice 1 voip

description **Incoming Call from SIP Trunk**

translation-profile incoming FROM_SIP

session protocol sipv2

session target sip-server

incoming called-number .

voice-class codec 1 

!

telephony-service

max-ephones 2

max-dn 2

ip source-address 172.30.0.73 port 2000

max-conferences 8 gain -6

transfer-system full-consult

create cnf-files version-stamp 7960 Nov 28 2011 17:09:42

!

!

ephone-dn  1

number 1412

!

!

ephone  1

mac-address 0018.B956.26A6

type 7940

button  1:1

!

Here logs-

Nov 28 14:06:30.855: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

INVITE sip:310814@10.225.73.142:5060;user=phone SIP/2.0

Accept: application/sdp;q=0.3

Accept: application/ISUP;q=0.2

Accept: multipart/mixed;q=0.1

Allow: INVITE,ACK,CANCEL,BYE,INFO,PRACK,UPDATE,OPTIONS,REGISTER,REFER,SUBSCRIBE

Call-ID: 41bdd18rhx0n3ad9j@10.225.65.10

Contact: "88123326563" <sip:88123326563@10.225.65.10:5060;user=phone>

CSeq: 195 INVITE

Expires: 3600

From: "88123326563" <sip:88123326563@10.225.65.10:5060;user=phone>;tag=7nuqb9vo60

To: "310814" <sip:310814@10.225.73.142:5060;user=phone>

Organization: IskraTel

Supported: 100rel

User-Agent: SI3000

Via: SIP/2.0/UDP 10.225.65.10:5060;branch=z9hG4bK-eu09i-y9dus

Max-Forwards: 69

Subject: Call from CS6111

Content-Length:  243

Content-Type: application/sdp

Content-Disposition: session;handling=required

v=0

o=- 1381660 7688784 IN IP4 10.225.65.111

s=-

c=IN IP4 10.225.65.111

b=AS:64

t=0 0

m=audio 24802 RTP/AVP 8 0 101

a=rtpmap:8 PCMA/8000

a=rtpmap:0 PCMU/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=ptime:20

a=sendrecv

Nov 28 14:06:30.859: //42/017E7BA28057/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 10.225.65.10:5060;branch=z9hG4bK-eu09i-y9dus

From: "88123326563" <sip:88123326563@10.225.65.10:5060;user=phone>;tag=7nuqb9vo60

To: "310814" <sip:310814@10.225.73.142:5060;user=phone>

Date: Mon, 28 Nov 2011 14:06:30 GMT

Call-ID: 41bdd18rhx0n3ad9j@10.225.65.10

CSeq: 195 INVITE

Allow-Events: telephone-event

Server: Cisco-SIPGateway/IOS-12.x

Content-Length: 0

Nov 28 14:06:30.859: //42/017E7BA28057/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 403 Forbidden

Via: SIP/2.0/UDP 10.225.65.10:5060;branch=z9hG4bK-eu09i-y9dus

From: "88123326563" <sip:88123326563@10.225.65.10:5060;user=phone>;tag=7nuqb9vo60

To: "310814" <sip:310814@10.225.73.142:5060;user=phone>;tag=1FA29624-172

Date: Mon, 28 Nov 2011 14:06:30 GMT

Call-ID: 41bdd18rhx0n3ad9j@10.225.65.10

CSeq: 195 INVITE

Allow-Events: telephone-event

Server: Cisco-SIPGateway/IOS-12.x

Reason: Q.850;cause=21

Content-Length: 0

Nov 28 14:06:30.859: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

ACK sip:310814@10.225.73.142:5060;user=phone SIP/2.0

Via: SIP/2.0/UDP 10.225.65.10:5060;branch=z9hG4bK-eu09i-y9dus

From: "88123326563" <sip:88123326563@10.225.65.10:5060;user=phone>;tag=7nuqb9vo60

To: "310814" <sip:310814@10.225.73.142:5060;user=phone>;tag=1FA29624-172

Call-ID: 41bdd18rhx0n3ad9j@10.225.65.10

CSeq: 195 ACK

Max-Forwards: 70

Content-Length: 0

1 ACCEPTED SOLUTION

Accepted Solutions
Chris Deren
Hall of Fame Master

Looking at your debug closly I see the following:

SIP/2.0 403 Forbidden

Do you have SIP trunk authentication proerply set?  This would be under sip-ua config, or under the dial-peer pointing to SIP trunk.

Also, if this IOS is 15+ make sure your default toll-fraud security feature is properly showing expected allowed connections. For test you can add the IP addresses this way:

Router(config)#voice service voip

Router(conf-voi-serv)#ip address trusted list

Router(cfg-iptrust-list)#ipv4 x.x.x.x 255.255.255.255

Chris

View solution in original post

7 REPLIES 7
Chris Deren
Hall of Fame Master

Can you post "debug voice translation" and "debug voice dialpeer"?

Chris

Hi!

Thank you for reply!

I see "No Outgoing Dial-peer Is Matched"

Do I need outgoing dial-peer if I have registered phone on CME?

Here my debug-

Log Buffer (5120 bytes):

o Type=DIALPEER_INFO_SPEECH

Nov 28 18:09:28.090: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

љљ Match Rule=DP_MATCH_DEST; Called Number=310817

Nov 28 18:09:28.090: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

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

Nov 28 18:09:28.090: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:

љљ dialstring=310817, saf_enabled=1, saf_dndb_lookup=1, dp_result=-1

Nov 28 18:09:28.090: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersMoreArg:

љљ Result=NO_MATCH(-1)

Nov 28 18:09:28.090: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:

љљ Calling Number=89653815299, Called Number=, Voice-Interface=0x0,

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

љљ Peer Info Type=DIALPEER_INFO_SPEECH

Nov 28 18:09:28.090: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:

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

Nov 28 18:09:28.090: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:

љљ dialstring=NULL, saf_enabled=0, saf_dndb_lookup=0, dp_result=-1

Nov 28 18:09:28.090: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:

љљ Calling Number=89653815299, Called Number=, Voice-Interface=0x0,

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

љљ Peer Info Type=DIALPEER_INFO_SPEECH

Nov 28 18:09:28.090: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:

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

Nov 28 18:09:28.090: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:

љљ dialstring=NULL, saf_enabled=0, saf_dndb_lookup=0, dp_result=-1

Nov 28 18:09:28.090: //-1/F234076480D5/DPM/dpAssociateIncomingPeerCore:

љљ Calling Number=89653815299, Called Number=310817, Voice-Interface=0x0,

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

љљ Peer Info Type=DIALPEER_INFO_SPEECH

Nov 28 18:09:28.090: //-1/F234076480D5/DPM/dpAssociateIncomingPeerCore:

љљ Result=Success(0) after DP_MATCH_INCOMING_DNIS; Incoming Dial-peer=1

Nov 28 18:09:28.090: //-1/F234076480D5/DPM/dpMatchSafModulePlugin:

љљ dialstring=NULL, saf_enabled=0, saf_dndb_lookup=0, dp_result=0

Nov 28 18:09:28.094: //-1/F234076480D5/RXRULE/regxrule_stack_pop_RegXruleNumInfo: stack=0x3111A7B8; count=1

Nov 28 18:09:28.094: //-1/F234076480D5/RXRULE/regxrule_stack_pop_callinfo_internal: numinfo=0x3114CF60

Nov 28 18:09:28.094: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

љљ Calling Number=, Called Number=310817, Peer Info Type=DIALPEER_INFO_SPEECH

Nov 28 18:09:28.094: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

љљ Match Rule=DP_MATCH_DEST; Called Number=310817

Nov 28 18:09:28.094: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

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

Nov 28 18:09:28.094: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:

љљ dialstring=310817, saf_enabled=0, saf_dndb_lookup=1, dp_result=-1

Nov 28 18:09:28.094: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersMoreArg:

љљ Result=NO_MATCH(-1)

Nov 28 18:09:28.094: //-1/F234076480D5/RXRULE/regxrule_stack_push_RegXruleNumInfo_internal: stack=0x3111A7B8; count=1

Nov 28 18:09:28.094: //-1/F234076480D5/RXRULE/regxrule_profile_translate_internal: number=89653815299 type=unknown plan=unknown numbertype=calling

Nov 28 18:09:28.094: //-1/F234076480D5/RXRULE/regxrule_get_RegXrule: Invalid translation ruleset tag=0

Nov 28 18:09:28.094: //-1/F234076480D5/RXRULE/regxrule_profile_match_internal: Error: ruleset for calling number not found

Nov 28 18:09:28.094: //-1/F234076480D5/RXRULE/regxrule_profile_translate_internal: No match: number=89653815299 type=unknown plan=unknown

Nov 28 18:09:28.094: //-1/F234076480D5/RXRULE/regxrule_profile_translate_internal: number= type=unknown plan=unknown numbertype=redirect-called

Nov 28 18:09:28.094: //-1/F234076480D5/RXRULE/regxrule_get_RegXrule: Invalid translation ruleset tag=0

Nov 28 18:09:28.094: //-1/F234076480D5/RXRULE/regxrule_profile_match_internal: Error: ruleset for redirect-called number not found

Nov 28 18:09:28.094: //-1/F234076480D5/RXRULE/regxrule_profile_translate_internal: No match: number= type=unknown plan=unknown

Nov 28 18:09:28.094: //-1/F234076480D5/RXRULE/regxrule_profile_translate_internal: number=310817 type=unknown plan=unknown numbertype=called

Nov 28 18:09:28.094: //-1/F234076480D5/RXRULE/regxrule_profile_match_internal: Matched with rule 1 in ruleset 1

Nov 28 18:09:28.094: //-1/F234076480D5/RXRULE/regxrule_profile_match_internal: Matched with rule 1 in ruleset 1

Nov 28 18:09:28.094: //-1/F234076480D5/RXRULE/sed_subst: Successful substitution; pattern=310817 matchPattern=310817 replacePattern=1412 replaced pattern=1412

Nov 28 18:09:28.094: //-1/F234076480D5/RXRULE/regxrule_subst_num_type: Match Type = none, Replace Type = none Input Type = unknown

Nov 28 18:09:28.094: //-1/F234076480D5/RXRULE/regxrule_subst_num_plan: Match Plan = none, Replace Plan = none Input Plan = unknown

Nov 28 18:09:28.094: //-1/F234076480D5/RXRULE/regxrule_profile_translate_internal: xlt_number=1412 xlt_type=unknown xlt_plan=unknown

Nov 28 18:09:28.098: //-1/F234076480D5/RXRULE/regxrule_stack_pop_RegXruleNumInfo: stack=0x3111A7B8; count=1

Nov 28 18:09:28.098: //-1/F234076480D5/RXRULE/regxrule_stack_pop_callinfo_internal: numinfo=0x3114CF60

Chris Deren
Hall of Fame Master

I always build one with session target of the IP assigned as ip source-address in your case 172.30.0.73, it does not hurt to try it.

HTH,

Chris

I did in another way.

Is it correct?

dial-peer voice 1 voip

љdescription **Incoming Call from SIP Trunk**

љsession protocol sipv2

љsession target sip-server

љincoming called-number .

љvoice-class codec 1љ

!

dial-peer voice 2 voip

љdestination-pattern 1...

љsession target ipv4:172.30.0.73

љvoice-class codec 1љ

!

!

num-exp 310817 1412

 

 

Nov 28 19:31:26.510: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

љљ Calling Number=310817, Called Number=310817, Peer Info Type=DIALPEER_INFO_SPEECH

Nov 28 19:31:26.510: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

љљ Match Rule=DP_MATCH_DEST; Called Number=310817

Nov 28 19:31:26.510: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

љљ Result=Success(0) after DP_MATCH_DEST

Nov 28 19:31:26.510: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:

љљ dialstring=310817, saf_enabled=1, saf_dndb_lookup=1, dp_result=0

Nov 28 19:31:26.510: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersMoreArg:

љљ Result=SUCCESS(0)

љљ List of Matched Outgoing Dial-peer(s):

љљљљ 1: Dial-peer Tag=20001

љљљљ 2: Dial-peer Tag=2

Nov 28 19:31:26.510: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:

љљ Calling Number=89653815299, Called Number=, Voice-Interface=0x0,

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

љљ Peer Info Type=DIALPEER_INFO_SPEECH

Nov 28 19:31:26.510: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:

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

Nov 28 19:31:26.510: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:

љљ dialstring=NULL, saf_enabled=0, saf_dndb_lookup=0, dp_result=-1

Nov 28 19:31:26.510: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:

љљ Calling Number=89653815299, Called Number=, Voice-Interface=0x0,

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

љљ Peer Info Type=DIALPEER_INFO_SPEECH

Nov 28 19:31:26.510: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:

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

Nov 28 19:31:26.510: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:

љљ dialstring=NULL, saf_enabled=0, saf_dndb_lookup=0, dp_result=-1

Nov 28 19:31:26.510: //-1/65CF4E9C8117/DPM/dpAssociateIncomingPeerCore:

љљ Calling Number=89653815299, Called Number=310817, Voice-Interface=0x0,

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

љљ Peer Info Type=DIALPEER_INFO_SPEECH

Nov 28 19:31:26.510: //-1/65CF4E9C8117/DPM/dpAssociateIncomingPeerCore:

љљ Result=Success(0) after DP_MATCH_INCOMING_DNIS; Incoming Dial-peer=1

Nov 28 19:31:26.510: //-1/65CF4E9C8117/DPM/dpMatchSafModulePlugin:

љљ dialstring=NULL, saf_enabled=0, saf_dndb_lookup=0, dp_result=0

Nov 28 19:31:26.514: //-1/65CF4E9C8117/RXRULE/regxrule_stack_pop_RegXruleNumInfo: stack=0x3111A4A0; count=1

Nov 28 19:31:26.514: //-1/65CF4E9C8117/RXRULE/regxrule_stack_pop_callinfo_internal: numinfo=0x3114D1B0

Nov 28 19:31:26.514: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

љљ Calling Number=, Called Number=310817, Peer Info Type=DIALPEER_INFO_SPEECH

Nov 28 19:31:26.514: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

љљ Match Rule=DP_MATCH_DEST; Called Number=310817

Nov 28 19:31:26.514: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

љљ Result=Success(0) after DP_MATCH_DEST

Nov 28 19:31:26.514: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:

љљ dialstring=310817, saf_enabled=0, saf_dndb_lookup=1, dp_result=0

Nov 28 19:31:26.514: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersMoreArg:

љљ Result=SUCCESS(0)

љљ List of Matched Outgoing Dial-peer(s):

љљљљ 1: Dial-peer Tag=20001

љљљљ 2: Dial-peer Tag=2

Nov 28 19:31:26.514: //-1/65CF4E9C8117/RXRULE/regxrule_stack_push_RegXruleNumInfo_internal: stack=0x3111A4A0; count=1

Nov 28 19:31:26.518: //-1/65CF4E9C8117/RXRULE/regxrule_stack_pop_RegXruleNumInfo: stack=0x3111A4A0; count=1

Nov 28 19:31:26.518: //-1/65CF4E9C8117/RXRULE/regxrule_stack_pop_callinfo_internal: numinfo=0x3114D1B0

Forgot to add-the call is failed

Chris Deren
Hall of Fame Master

Looking at your debug closly I see the following:

SIP/2.0 403 Forbidden

Do you have SIP trunk authentication proerply set?  This would be under sip-ua config, or under the dial-peer pointing to SIP trunk.

Also, if this IOS is 15+ make sure your default toll-fraud security feature is properly showing expected allowed connections. For test you can add the IP addresses this way:

Router(config)#voice service voip

Router(conf-voi-serv)#ip address trusted list

Router(cfg-iptrust-list)#ipv4 x.x.x.x 255.255.255.255

Chris

View solution in original post

Thank you!

it's hepled-

Router(config)#voice service voip

Router(conf-voi-serv)#ip address trusted list

Router(cfg-iptrust-list)#ipv4 x.x.x.x 255.255.255.255

Best regards!

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