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New Member

cisco 2900 sip trunk

Im trying to configure a sip trunk with my isp on a 2900 series. Need help

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Hall of Fame Super Silver

cisco 2900 sip trunk

Add the following:

voice service voip

no ip address trusted authenticate

Chris

18 REPLIES
Hall of Fame Super Gold

cisco 2900 sip trunk

Maybe hire a reputable consultant or certified partner ?

New Member

cisco 2900 sip trunk

New Member

cisco 2900 sip trunk

Timon i did try that configuratio example and im familiar with cisco voice configurations. but im not understanding the problem with the connection to the isp since no packets are being sent when incoming call is sent to my router

New Member

Re: cisco 2900 sip trunk

is your router registered on ISP registrar?

NAT, Firewall settings are correct?

Give us more informations, Post some debugs

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Hall of Fame Super Silver

Re: cisco 2900 sip trunk

What iOS version are you using? You may be running into VoIP authentication feature blocking the traffic.

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New Member

Re: cisco 2900 sip trunk

ios 15.1, i managed to make outgoing calls by fixing config errors ondial peers and translation rules. Now i created a translation rule for incomig calls using called number and i dont get any calls outside the trunk ( fast busy signal)

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Hall of Fame Super Silver

Re: cisco 2900 sip trunk

Please post your config.

Chris

New Member

Re: cisco 2900 sip trunk

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

h323

sip

  bind control source-interface GigabitEthernet0/1.208

  bind media source-interface GigabitEthernet0/1.208

  registrar server

!

voice class codec 1

codec preference 1 g711ulaw

!

!

voice register global

mode cme

source-address 172.20.100.1 port 5060

max-dn 30

max-pool 30

create profile sync 0065402769141141

voice register pool  13

id mac 2C41.3816.6880

number 1 dn 13

preference 1

no call-waiting

codec g711alaw

!

!

!

voice translation-rule 1

rule 1 /35552901060/ /140/

!

voice translation-rule 3

rule 1 /35552901060/ /150/

!

voice translation-rule 10

rule 1 /^.*/ /35552901060/

!

voice translation-rule 101

rule 1 /^0\(.........\)/ /355\1/

!

!

voice translation-profile CEL-DNIS-XLATE

translate called 101

!

voice translation-profile INTERNATIONAL-TP

translate called 1001

!

voice translation-profile SIP-IN

translate called 3

!

voice translation-profile inboundTrunk

translate called 1

interface GigabitEthernet0/0.100

description "Voice VLAN Interface"

encapsulation dot1Q 100

ip address 172.20.100.1 255.255.255.0

dial-peer cor custom

name TIRANA

name INTERURBANE

name MOBILE

name INTERNATIONAL

!

!

dial-peer cor list NO-ACCESS

!

dial-peer cor list CALL-TIRANA

member TIRANA

!

dial-peer cor list CALL-INTERURBANE

member INTERURBANE

!

dial-peer cor list CALL-MOBILE

member MOBILE

!

dial-peer cor list CALL-INTERNATIONAL

member INTERNATIONAL

!

!

dial-peer voice 2 voip

description *** Incoming call to  - -- Generic -- - SIP Trunk ***

destination-pattern 0T

session protocol sipv2

session target sip-server

incoming called-number .T

dtmf-relay sip-notify

codec g729br8

!

dial-peer voice 20 voip

translation-profile incoming SIP-IN

redirect ip2ip

session target sip-server

incoming called-number .%

codec g729br8

!

dial-peer voice 1 voip

description incoming SIP Trunk

translation-profile incoming SIP-IN

redirect ip2ip

translate-outgoing calling 10

incoming called-number 35552901060

codec g729br8

!

!

sip-ua

credentials username xxxxxxx password 7 xxxxxxx realm xxxxxx

authentication username xxxxxxxxxxxx password 7 xxxxxxx

calling-info pstn-to-sip from number set 35552901060

no remote-party-id

timers connect 100

registrar ipv4:80.78.66.70 expires 3600

sip-server ipv4:80.78.66.70

!

!

!

gatekeeper

shutdown

!

!

telephony-service

max-ephones 24

max-dn 30

ip source-address 172.20.100.1 port 2000

max-conferences 8 gain -6

transfer-system full-consult

create cnf-files version-stamp 7960 Jan 11 2012 18:15:56

part of voice config and sip trunk

New Member

Re: cisco 2900 sip trunk

debug ccsip  calls messages

SIP Call statistics tracing is enabled

#

Jan 17 20:27:43.311: //13240/571F11647758/SIP/Call/sipSPICallInfo:

The Call Setup Information is:

Call Control Block (CCB) : 0x2CD73768

State of The Call        : STATE_DEAD

TCP Sockets Used         : NO

Calling Number           :

Called Number            : 35552901060

Source IP Address (Sig  ): 192.168.207.77

Destn SIP Req Addr:Port  : 80.78.66.70:5060

Destn SIP Resp Addr:Port : 80.78.66.70:5060

Destination Name         : 80.78.66.70

Jan 17 20:27:43.311: //13240/571F11647758/SIP/Call/sipSPIMediaCallInfo:

Number of Media Streams: 1

Media Stream             : 1

Negotiated Codec         : g729br8

Negotiated Codec Bytes   : 20

Nego. Codec payload      : 18 (tx), 18 (rx)

Negotiated Dtmf-relay    : 0

Dtmf-relay Payload       : 0 (tx), 0 (rx)

Source IP Address (Media): 192.168.207.77

Source IP Port    (Media): 25666

Destn  IP Address (Media): 80.78.64.16

Destn  IP Port    (Media): 21338

Orig Destn IP Address:Port (Media): [ - ]:0

Jan 17 20:27:43.311: //13240/571F11647758/SIP/Call/sipSPICallInfo:

Disconnect Cause (CC)    : 21

Disconnect Cause (SIP)   : 403

Jan 17 20:27:43.655: //13241/B7806B4D174E/SIP/Call/sipSPICallInfo:

The Call Setup Information is:

Call Control Block (CCB) : 0x2CD73768

State of The Call        : STATE_DEAD

TCP Sockets Used         : NO

Calling Number           :

Called Number            : 35552901060

Source IP Address (Sig  ): 192.168.207.77

Destn SIP Req Addr:Port  : 80.78.66.70:5060

Destn SIP Resp Addr:Port : 80.78.66.70:5060

Destination Name         : 80.78.66.70

Jan 17 20:27:43.655: //13241/B7806B4D174E/SIP/Call/sipSPIMediaCallInfo:

Number of Media Streams: 1

Media Stream             : 1

Negotiated Codec         : g729br8

Negotiated Codec Bytes   : 20

Nego. Codec payload      : 18 (tx), 18 (rx)

Negotiated Dtmf-relay    : 0

Dtmf-relay Payload       : 0 (tx), 0 (rx)

Source IP Address (Media): 192.168.207.77

Source IP Port    (Media): 17616

Destn  IP Address (Media): 80.78.64.16

Destn  IP Port    (Media): 21340

Orig Destn IP Address:Port (Media): [ - ]:0

Jan 17 20:27:43.655: //13241/B7806B4D174E/SIP/Call/sipSPICallInfo:

Disconnect Cause (CC)    : 21

Disconnect Cause (SIP)   : 403

Hall of Fame Super Silver

cisco 2900 sip trunk

Add the following:

voice service voip

no ip address trusted authenticate

Chris

New Member

cisco 2900 sip trunk

after adding the line the phone company says the dialed number is not correct

Hall of Fame Super Silver

cisco 2900 sip trunk

Can you post the SIP debug for the call?

is your epone-dn defined as DN 150?

Can you also post debug voice dial-peer?

Chris

New Member

cisco 2900 sip trunk

ephone-dn  1

number 150

label koli

ephone  1

device-security-mode none

mac-address 0005.9A3C.7800

type CIPC

button  1:1

debug ccsip

Jan 17 20:51:30.799: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

   Calling Number=35552901060, Called Number=35552901060, Peer Info Type=DIALPEER_INFO_SPEECH

Jan 17 20:51:30.799: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

   Match Rule=DP_MATCH_DEST; Called Number=35552901060

Jan 17 20:51:30.799: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

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

Jan 17 20:51:30.799: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:

   dialstring=35552901060, saf_enabled=1, saf_dndb_lookup=1, dp_result=-1

Jan 17 20:51:30.799: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersMoreArg:

   Result=NO_MATCH(-1)

Jan 17 20:51:30.799: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:

   Calling Number=, 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 17 20:51:30.799: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:

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

Jan 17 20:51:30.799: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:

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

Jan 17 20:51:30.799: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:

   Calling Number=, 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 17 20:51:30.799: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:

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

Jan 17 20:51:30.799: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:

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

Jan 17 20:51:30.799: //-1/A6E9DC6E76CD/DPM/dpAssociateIncomingPeerCore:

   Calling Number=, Called Number=35552901060, Voice-Interface=0x0,

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

   Peer Info Type=DIALPEER_INFO_SPEECH

Jan 17 20:51:30.799: //-1/A6E9DC6E76CD/DPM/dpAssociateIncomingPeerCore:

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

Jan 17 20:51:30.799: //-1/A6E9DC6E76CD/DPM/dpMatchSafModulePlugin:

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

Jan 17 20:51:30.803: //-1/A6E9DC6E76CD/DPM/dpMatchPeersCore:

   Calling Number=, Called Number=150, Peer Info Type=DIALPEER_INFO_SPEECH

Jan 17 20:51:30.803: //-1/A6E9DC6E76CD/DPM/dpMatchPeersCore:

   Match Rule=DP_MATCH_DEST; Called Number=150

Jan 17 20:51:30.803: //-1/A6E9DC6E76CD/DPM/dpMatchPeersCore:

   Result=Success(0) after DP_MATCH_DEST

Jan 17 20:51:30.803: //-1/A6E9DC6E76CD/DPM/dpMatchSafModulePlugin:

   dialstring=150, saf_enabled=0, saf_dndb_lookup=1, dp_result=0

Jan 17 20:51:30.803: //-1/A6E9DC6E76CD/DPM/dpMatchPeersMoreArg:

   Result=SUCCESS(0)

   List of Matched Outgoing Dial-peer(s):

     1: Dial-peer Tag=20002

Jan 17 20:51:30.803: //-1/A6E9DC6E76CD/DPM/dpMatchPeersCore:

   Calling Number=, Called Number=150, Peer Info Type=DIALPEER_INFO_SPEECH

Jan 17 20:51:30.803: //-1/A6E9DC6E76CD/DPM/dpMatchPeersCore:

   Match Rule=DP_MATCH_DEST; Called Number=150

Jan 17 20:51:30.803: //-1/A6E9DC6E76CD/DPM/dpMatchPeersCore:

   Result=Success(0) after DP_MATCH_DEST

Jan 17 20:51:30.803: //-1/A6E9DC6E76CD/DPM/dpMatchSafModulePlugin:

   dialstring=150, saf_enabled=0, saf_dndb_lookup=1, dp_result=0

Jan 17 20:51:30.803: //-1/A6E9DC6E76CD/DPM/dpMatchPeersMoreArg:

   Result=SUCCESS(0)

   List of Matched Outgoing Dial-peer(s):

     1: Dial-peer Tag=20002

Jan 17 20:51:30.803: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

   Calling Number=150, Called Number=150, Peer Info Type=DIALPEER_INFO_SPEECH

Jan 17 20:51:30.803: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

   Match Rule=DP_MATCH_DEST; Called Number=150

Jan 17 20:51:30.803: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

   Result=Success(0) after DP_MATCH_DEST

Jan 17 20:51:30.803: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:

   dialstring=150, saf_enabled=0, saf_dndb_lookup=1, dp_result=0

Jan 17 20:51:30.803: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersMoreArg:

   Result=SUCCESS(0)

   List of Matched Outgoing Dial-peer(s):

     1: Dial-peer Tag=20002

Jan 17 20:51:30.803: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:

   Calling Number=150, 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 17 20:51:30.803: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:

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

Jan 17 20:51:30.807: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:

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

Jan 17 20:51:30.807: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:

   Calling Number=150, 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 17 20:51:30.807: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:

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

Jan 17 20:51:30.807: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:

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

Jan 17 20:51:30.807: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

   Calling Number=, Called Number=150, Peer Info Type=DIALPEER_INFO_SPEECH

Jan 17 20:51:30.807: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

   Match Rule=DP_MATCH_DEST; Called Number=150

Jan 17 20:51:30.807: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

   Result=Success(0) after DP_MATCH_DEST

Jan 17 20:51:30.807: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:

   dialstring=150, saf_enabled=0, saf_dndb_lookup=1, dp_result=0

Jan 17 20:51:30.807: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersMoreArg:

   Result=SUCCESS(0)

   List of Matched Outgoing Dial-peer(s):

     1: Dial-peer Tag=20002

Jan 17 20:51:30.807: //-1/A6E9DC6E76CD/DPM/dpMatchPeersCore:

   Calling Number=, Called Number=150, Peer Info Type=DIALPEER_INFO_SPEECH

Jan 17 20:51:30.807: //-1/A6E9DC6E76CD/DPM/dpMatchPeersCore:

   Match Rule=DP_MATCH_DEST; Called Number=150

Jan 17 20:51:30.807: //-1/A6E9DC6E76CD/DPM/dpMatchPeersCore:

   Result=Success(0) after DP_MATCH_DEST

Jan 17 20:51:30.807: //-1/A6E9DC6E76CD/DPM/dpMatchSafModulePlugin:

   dialstring=150, saf_enabled=1, saf_dndb_lookup=1, dp_result=0

Jan 17 20:51:30.807: //-1/A6E9DC6E76CD/DPM/dpMatchPeersMoreArg:

   Result=SUCCESS(0)

   List of Matched Outgoing Dial-peer(s):

     1: Dial-peer Tag=20002

Jan 17 20:51:30.811: //13290/A6E9DC6E76CD/SIP/Call/sipSPICallInfo:

Hall of Fame Super Silver

cisco 2900 sip trunk

Looks like dial-peer matching is working and the dial peer for the ephon-dn is located.

Are you still getting "Disconnect Cause (SIP)   : 403" in the sip debug? Can you post latest debug for this call?

New Member

cisco 2900 sip trunk

now its 404

The Call Setup Information is:

Call Control Block (CCB) : 0x2CD73768

State of The Call        : STATE_DEAD

TCP Sockets Used         : NO

Calling Number           :

Called Number            : 35552901060

Source IP Address (Sig  ): 192.168.207.77

Destn SIP Req Addr:Port  : 80.78.66.70:5060

Destn SIP Resp Addr:Port : 80.78.66.70:5060

Destination Name         : 80.78.66.70

Jan 17 21:07:11.955: //13323/B2DC8DC79CDF/SIP/Call/sipSPIMediaCallInfo:

Number of Media Streams: 1

Media Stream             : 1

Negotiated Codec         : g729br8

Negotiated Codec Bytes   : 20

Nego. Codec payload      : 18 (tx), 18 (rx)

Negotiated Dtmf-relay    : 0

Dtmf-relay Payload       : 0 (tx), 0 (rx)

Source IP Address (Media): 192.168.207.77

Source IP Port    (Media): 24754

Destn  IP Address (Media): 80.78.64.16

Destn  IP Port    (Media): 23024

Orig Destn IP Address:Port (Media): [ - ]:0

Jan 17 21:07:11.955: //13323/B2DC8DC79CDF/SIP/Call/sipSPICallInfo:

Disconnect Cause (CC)    : 1

Disconnect Cause (SIP)   : 404

Hall of Fame Super Silver

cisco 2900 sip trunk

404 indicates unallocated number, so the 150 extnesion does not appear to be found. I see you have cor lists defined, but since you did not post the full config, can it be that COR list is blocking the call?

Not, that it matters here, but why are you using g729br8 codec, and not g729r8?

New Member

Re: cisco 2900 sip trunk

if i change codec ?

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New Member

Re: cisco 2900 sip trunk

No the corlist is not blocking anything on any dial peer

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