Solved: Re: SIP trunk to provider

rbblue234 · ‎03-11-2013

Hi All,

Currently using a cisco router as my SIP GW. My hope is to use this router as my sip trunk to my provider. When i call the number assigned to the gw im getting the following two entries in the debug.

//12/0E601AFD8035/SIP/Error/sipSPI_ipip_copy_sdp_to_channelInfo: failed to update call entry

//-1/xxxxxxxxxxxx/SIP/Error/sipSPIGetContentQ931: No Inbound Container Created !!!

below is my run.

=~=~=~=~=~=~=~=~=~=~=~= PuTTY log 2013.03.11 20:15:33 =~=~=~=~=~=~=~=~=~=~=~=

Using username "gwAdmin".

Using keyboard-interactive authentication.

Password:

VoIPGw>en

Password:

VoIPGw#sh run

Building configuration...

Current configuration : 3271 bytes

!

version 12.4

no service pad

service timestamps debug datetime msec

service timestamps log datetime msec

no service password-encryption

!

hostname VoIPGw

!

boot-start-marker

boot-end-marker

!

logging message-counter syslog

enable password Mansoor91111

!

aaa new-model

!

aaa session-id common

--More-- !

!

dot11 syslog

ip source-route

ip cef

!

ip domain name VoIPGw

ip name-server 8.8.8.8

!

no ipv6 cef

!

multilink bundle-name authenticated

!

voice service voip

allow-connections sip to sip

redirect ip2ip

sip

bind control source-interface FastEthernet0/0

--More-- bind media source-interface FastEthernet0/0

header-passing

sip-profiles 1

!

voice class codec 1

codec preference 1 g729r8

codec preference 2 g711ulaw

!

voice class sip-profiles 1

response ANY sip-header Allow-Header modify "invite, options, bye, cancel, ack, prack, update, refer, subscribe, notify, info, register" "invite, bye, cancel, ack, prack, subscribe, notify, info, register"

response 183 sip-header Allow-Header modify "ACK, BYE, CANCEL, INFO, INVITE, OPTIONS, PRACK, REFER, NOTIFY" "ACK, BYE, CANCEL, INFO, INVITE, PRACK, refer, notify"

response ANY sip-header Allow-Header modify "INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER" "INVITE, BYE, CANCEL"

response ANY sip-header Allow-Header modify "INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER" "INVITE, BYE, CANCEL, ACK, PRACK, SUBSCRIBE, NOTIFY, INFO, REGISTER"

--More-- !

!

voice-card 0

no dspfarm

!

username gwAdmin password 0 Mansoor91111

!

archive

log config

hidekeys

!

--More-- !

ip ssh time-out 60

ip ssh authentication-retries 2

!

interface FastEthernet0/0

ip address <removed> 255.255.255.240

duplex auto

speed auto

!

interface Integrated-Service-Engine0/0

no ip address

shutdown

!

interface FastEthernet0/1/0

!

interface FastEthernet0/1/1

shutdown

!

interface FastEthernet0/1/2

!

interface FastEthernet0/1/3

--More-- !

interface FastEthernet0/1/4

!

interface FastEthernet0/1/5

!

interface FastEthernet0/1/6

!

interface FastEthernet0/1/7

!

interface FastEthernet0/1/8

!

interface Vlan1

description MGMT Vlan

ip address 192.168.10.52 255.255.255.0

!

interface Vlan4

description Server Vlan

no ip address

!

ip default-gateway <removed>

ip forward-protocol nd

ip route 0.0.0.0 0.0.0.0 FastEthernet0/0

!

--More-- no ip http server

no ip http secure-server

!

control-plane

!

voice-port 0/0/0

!

voice-port 0/0/1

!

voice-port 0/0/2

!

voice-port 0/0/3

!

voice-port 0/1/0

!

--More-- voice-port 0/1/1

!

voice-port 0/1/2

!

voice-port 0/1/3

!

voice-port 0/4/0

auto-cut-through

signal immediate

input gain auto-control

description Music On Hold Port

!

dial-peer voice 100 voip

destination-pattern <removed>

session target ipv4:192.168.10.51

session transport udp

dtmf-relay rtp-nte

no vad

!

sip-ua

--More-- credentials username 138957_yup password 7 10620C0D1613423D5C0D3A6565 realm 138957_yup

authentication username 138957_yup password 7 10620C0D1613423D5C0D3A6565

registrar dns:newyork.voip.ms expires 3600

!

line con 0

no modem enable

line aux 0

line 2

no activation-character

no exec

transport preferred none

transport input all

line vty 0 4

password cisco

transport input ssh

!

end

VoIPGw# exit

Ayodeji Okanlawon · ‎03-12-2013

Is this the dial-peer you are using to send calls to your ITSP

dial-peer voice 100 voip

destination-pattern

session target ipv4:192.168.10.51

session transport udp

dtmf-relay rtp-nte

no vad

It is not configured correctly..You need to add

session protocol sipv2

Add that do another test and send

debug ccsip messages

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View solution in original post

Ayodeji Okanlawon · ‎03-12-2013

Is this the dial-peer you are using to send calls to your ITSP

dial-peer voice 100 voip

destination-pattern

session target ipv4:192.168.10.51

session transport udp

dtmf-relay rtp-nte

no vad

It is not configured correctly..You need to add

session protocol sipv2

Add that do another test and send

debug ccsip messages

Please rate all useful posts

"opportunity is a haughty goddess who waste no time with those who are unprepared"

Please rate all useful posts

rbblue234 · ‎03-12-2013

DOH! Clicked the wrong button

The dial peer im using is trying too to deliver inbound calls to CUCM.

rbblue234 · ‎03-12-2013

below is my updated dial peer config with sip messages.

dial-peer voice 100 voip

destination-pattern

voice-class sip options-ping 60

session protocol sipv2

session target ipv4:192.168.10.51

session transport udp

dtmf-relay rtp-nte

VoIPGw#

*Mar 12 12:23:07.697: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

INVITE sip:XXXXXX7522@XX.XX.134.24:5060 SIP/2.0

Via: SIP/2.0/UDP 74.63.41.218:5060;branch=z9hG4bK2d8e8860

Max-Forwards: 70

From: "GARRETT " ;tag=as587c6f6e

To: <>XXXXXX7522@XX.XX.134.24:5060>

Contact:

Call-ID: 5313cd90691bd5cd3a12d9b76bdeaaf7@74.63.41.218:5060

CSeq: 102 INVITE

User-Agent: voip.ms

Date: Tue, 12 Mar 2013 11:39:59 GMT

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

Supported: replaces, timer

Remote-Party-ID: "GARRETT " ;party=calling;privacy=off;screen=no

Content-Type: application/sdp

Content-Length: 268

v=0

o=root 615418521 615418521 IN IP4 74.63.41.218

s=voip.ms

c=IN IP4 74.63.41.218

t=0 0

m=audio 15710 RTP/AVP 0 18 101

a=rtpmap:0 PCMU/8000

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=sendrecv

*Mar 12 12:23:07.713: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 74.63.41.218:5060;branch=z9hG4bK2d8e8860

From: "GARRETT " ;tag=as587c6f6e

To: <>XXXXXX7522@XX.XX.134.24:5060>

Date: Tue, 12 Mar 2013 12:23:07 GMT

Call-ID: 5313cd90691bd5cd3a12d9b76bdeaaf7@74.63.41.218:5060

CSeq: 102 INVITE

Allow-Events: telephone-event

Server: Cisco-SIPGateway/IOS-12.x

Content-Length: 0

*Mar 12 12:23:07.713: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

INVITE sip:XXXXXX7522@192.168.10.51:5060 SIP/2.0

Via: SIP/2.0/UDP XX.XX.134.24:5060;branch=z9hG4bK1F488

Remote-Party-ID: "GARRETT " <>XXXXXX1293@XX.XX.134.24>;party=calling;screen=no;privacy=off

From: "GARRETT " <>XXXXXX1293@newyork.voip.ms>;tag=28C918C-3B4

To:

Date: Tue, 12 Mar 2013 12:23:07 GMT

Call-ID: 6E47F727-8A4611E2-80D8EADC-2EFD56DE@XX.XX.134.24

Supported: 100rel,timer,resource-priority,replaces

Min-SE: 1800

Cisco-Guid: 1850090159-2319847906-2161306332-788354782

User-Agent: Cisco-SIPGateway/IOS-12.x

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

CSeq: 101 INVITE

Timestamp: 1363090987

Contact: <>XXXXXX1293@XX.XX.134.24:5060>

Expires: 180

Allow-Events: telephone-event

Max-Forwards: 69

Content-Type: application/sdp

Content-Disposition: session;handling=required

Content-Length: 294

v=0

o=CiscoSystemsSIP-GW-UserAgent 4747 6170 IN IP4 XX.XX.134.24

s=SIP Call

c=IN IP4 XX.XX.134.24

t=0 0

m=audio 17316 RTP/AVP 18 101 19

c=IN IP4 XX.XX.134.24

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=rtpmap:19 CN/8000

a=ptime:20

*Mar 12 12:23:07.721: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP XX.XX.134.24:5060;branch=z9hG4bK1F488

From: "GARRETT " <>XXXXXX1293@newyork.voip.ms>;tag=28C918C-3B4

To:

Date: Tue, 12 Mar 2013 11:39:59 GMT

Call-ID: 6E47F727-8A4611E2-80D8EADC-2EFD56DE@XX.XX.134.24

CSeq: 101 INVITE

Allow-Events: presence

Content-Length: 0

*Mar 12 12:23:07.721: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 503 Service Unavailable

Via: SIP/2.0/UDP XX.XX.134.24:5060;branch=z9hG4bK1F488

From: "GARRETT " <>XXXXXX1293@newyork.voip.ms>;tag=28C918C-3B4

To: ;tag=1166479469

Date: Tue, 12 Mar 2013 11:39:59 GMT

Call-ID: 6E47F727-8A4611E2-80D8EADC-2EFD56DE@XX.XX.134.24

CSeq: 101 INVITE

Allow-Events: presence

Warning: 399 cucm "Unable to find a device handler for the request received on port 5060 from XX.XX.134.24"

Content-Length: 0

*Mar 12 12:23:07.729: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

ACK sip:XXXXXX7522@192.168.10.51:5060 SIP/2.0

Via: SIP/2.0/UDP XX.XX.134.24:5060;branch=z9hG4bK1F488

From: "GARRETT " <>XXXXXX1293@newyork.voip.ms>;tag=28C918C-3B4

To: ;tag=1166479469

Date: Tue, 12 Mar 2013 12:23:07 GMT

Call-ID: 6E47F727-8A4611E2-80D8EADC-2EFD56DE@XX.XX.134.24

Max-Forwards: 70

CSeq: 101 ACK

Allow-Events: telephone-event

Content-Length: 0

*Mar 12 12:23:07.729: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 500 Internal Server Error

Via: SIP/2.0/UDP 74.63.41.218:5060;branch=z9hG4bK2d8e8860

From: "GARRETT " ;tag=as587c6f6e

To: <>XXXXXX7522@XX.XX.134.24:5060>;tag=28C919C-737

Date: Tue, 12 Mar 2013 12:23:07 GMT

Call-ID: 5313cd90691bd5cd3a12d9b76bdeaaf7@74.63.41.218:5060

CSeq: 102 INVITE

Allow-Events: telephone-event

Server: Cisco-SIPGateway/IOS-12.x

Reason: Q.850;cause=63

Content-Length: 0

*Mar 12 12:23:07.813: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

ACK sip:XXXXXX7522@XX.XX.134.24:5060 SIP/2.0

Via: SIP/2.0/UDP 74.63.41.218:5060;branch=z9hG4bK2d8e8860

Max-Forwards: 70

From: "GARRETT " ;tag=as587c6f6e

To: <>XXXXXX7522@XX.XX.134.24:5060>;tag=28C919C-737

Contact:

Call-ID: 5313cd90691bd5cd3a12d9b76bdeaaf7@74.63.41.218:5060

CSeq: 102 ACK

User-Agent: voip.ms

Content-Length: 0

Ayodeji Okanlawon · ‎03-12-2013

Do you have a sip trunk configured on callmanager pointing to the ip address of your gateway?

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"opportunity is a haughty goddess who waste no time with those who are unprepared"

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rbblue234 · ‎03-12-2013

Hello Aokanlawon.

Yes I do.

rbblue234 · ‎03-12-2013

im now getting a 503 error message with an error code of

Unable to find a device handler for the request received on port 5060 from

The ip address that's referenced in the error (removed from post) is the public IP address of the router.

So i went into CUCM and changes the IP address to be the public IP and still getting the same error.

rbblue234 · ‎03-12-2013

I removed the bind command in the router and the sip messages are now showing the local ip address going to CUCM.

rbblue234 · ‎03-12-2013

This has been fixed.

A few things i needed to change which is odd... but what ever at this point.

I needed to add vad to the dial peer, added sipv2 session in the dial peer, and clicked media termination point required on the sip trunk.

below is my running config.

VoIPGw#sh run

Building configuration...

Current configuration : 3588 bytes

!

version 12.4

no service pad

service timestamps debug datetime msec

service timestamps log datetime msec

no service password-encryption

!

hostname VoIPGw

!

boot-start-marker

boot-end-marker

!

logging message-counter syslog

enable password Mansoor91111

!

aaa new-model

!

aaa session-id common

--More-- !

!

dot11 syslog

ip source-route

ip cef

!

ip domain name VoIPGw

ip name-server 8.8.8.8

!

no ipv6 cef

!

multilink bundle-name authenticated

!

voice rtp send-recv

!

voice service voip

allow-connections sip to sip

fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711ulaw

sip

--More-- header-passing

early-offer forced

midcall-signaling passthru

g729 annexb-all

sip-profiles 1

!

voice class codec 1

codec preference 1 g729r8

codec preference 2 g711ulaw

!

voice class sip-profiles 1

response ANY sip-header Allow-Header modify "invite, options, bye, cancel, ack, prack, update, refer, subscribe, notify, info, register" "invite, bye, cancel, ack, prack, subscribe, notify, info, register"

response 183 sip-header Allow-Header modify "ACK, BYE, CANCEL, INFO, INVITE, OPTIONS, PRACK, REFER, NOTIFY" "ACK, BYE, CANCEL, INFO, INVITE, PRACK, refer, notify"

response ANY sip-header Allow-Header modify "INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER" "INVITE, BYE, CANCEL"

--More-- response ANY sip-header Allow-Header modify "INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER" "INVITE, BYE, CANCEL, ACK, PRACK, SUBSCRIBE, NOTIFY, INFO, REGISTER"

request ACK sdp-header Audio-Attribute modify "sendonly" "sendrecv"

!

voice-card 0

dspfarm

dsp services dspfarm

!

username gwAdmin password 0 Mansoor91111

!

--More-- !

archive

log config

hidekeys

!

ip ssh authentication-retries 2

!

interface FastEthernet0/0

ip address XX.XX.134.24 255.255.255.240

duplex auto

speed auto

!

interface Integrated-Service-Engine0/0

no ip address

shutdown

!

interface FastEthernet0/1/0

!

interface FastEthernet0/1/1

shutdown

--More-- !

interface FastEthernet0/1/2

!

interface FastEthernet0/1/3

!

interface FastEthernet0/1/4

!

interface FastEthernet0/1/5

!

interface FastEthernet0/1/6

!

interface FastEthernet0/1/7

!

interface FastEthernet0/1/8

!

interface Vlan1

description MGMT Vlan

ip address 192.168.10.52 255.255.255.0

!

interface Vlan4

description Server Vlan

no ip address

!

--More-- ip default-gateway XX.XX.134.17

ip forward-protocol nd

ip route 0.0.0.0 0.0.0.0 FastEthernet0/0

!

no ip http server

no ip http secure-server

!

control-plane

!

voice-port 0/0/0

!

voice-port 0/0/1

!

voice-port 0/0/2

!

--More-- voice-port 0/0/3

!

voice-port 0/1/0

!

voice-port 0/1/1

!

voice-port 0/1/2

!

voice-port 0/1/3

!

voice-port 0/4/0

auto-cut-through

signal immediate

input gain auto-control

description Music On Hold Port

!

dspfarm profile 2 transcode

codec g711ulaw

codec g711alaw

codec g729ar8

codec g729abr8

codec g729r8

--More-- maximum sessions 8

shutdown

!

dial-peer voice 100 voip

destination-pattern XXXXXX7522

voice-class sip options-ping 60

session protocol sipv2

session target ipv4:192.168.10.51

session transport udp

dtmf-relay rtp-nte

!

sip-ua

credentials username 138957_yup password 7 10620C0D1613423D5C0D3A6565 realm 138957_yup

authentication username 138957_yup password 7 10620C0D1613423D5C0D3A6565

registrar dns:newyork.voip.ms expires 3600

!

line con 0

no modem enable

--More-- line aux 0

line 2

no activation-character

no exec

transport preferred none

transport input all

line vty 0 4

password cisco

transport input ssh

!

end

Ayodeji Okanlawon · ‎03-12-2013

You dont need MTP...I suggest you dont use it..

FIrst of all you need to configure an inbound dial-peer from your ITSP

Eg..

dial-peer voice xx voip

session protocol sipv2

incoming called number xxx (where XXX = is your DDI/DN pattern)

dtmf relay rtp-nte

no vad

Try this and remove the MTP on the sip trunk

On the Bind..

I noticed that the gateway is asking CUCM to send response back to the 50.20.134.24 ip....Can CUCM connect to this IP from the local interface?

What Interface does CUCM use to talk to the CUBE? Which IP have you configured for your sip trunk..You should put your bind on that IP

From the traces...

INVITE sip:XXXXXX7522@192.168.10.51:5060 SIP/2.0

Via: SIP/2.0/UDP 50.20.134.24:5060

The CUBE is sending invite to CUCM on ip 192.168.10.51

Please rate all useful posts

"opportunity is a haughty goddess who waste no time with those who are unprepared"

Please rate all useful posts

rbblue234 · ‎03-12-2013

HI,

Removing the media term point still allows calls to be placed.

When i change my dial-peer to be as follows... the calls no longer route. The deubg shows that there is no dial-perr match

dial-peer voice 100 voip

voice-class sip options-ping 60

session protocol sipv2

session target ipv4:192.168.10.51

session transport udp

incoming called-number XXX9177522

dtmf-relay rtp-nte

Ayodeji Okanlawon · ‎03-12-2013

You need two dial-peers..I didnt say remove dial-peer 100..

You need an inbound and outbounbd dial-peer

Configure inbound dial-peer as follows

dial-peer voice 101 voip

voice-class sip options-ping 60

session protocol sipv2

session transport udp

incoming called-number XXX9177522

dtmf-relay rtp-nte

Configure outbound dial-peer as follows

dial-peer voice 100 voip

destination-pattern XXXXXX7522

voice-class sip options-ping 60

session protocol sipv2

session target ipv4:192.168.10.51

session transport udp

dtmf-relay rtp-nte

Please rate all useful posts

"opportunity is a haughty goddess who waste no time with those who are unprepared"

Please rate all useful posts

rbblue234 · ‎03-12-2013

I have to admit that im really having a hard time here...

With your config i can get calls comming in with out issues. I removed my programming and went directly with yours.

When i try and do an out bound call... well that failes.

I created the following dial peers. my CUCM will present 1+area code+ 7 digits

dial-peer voice 201 voip

voice-class sip options-ping 60

session protocol sipv2

session transport udp

incoming called-number 1.*

dtmf-relay rtp-nte

!

dial-peer voice 200 voip

destination-pattern 1.*

voice-class sip options-ping 60

session protocol sipv2

session target dns:neywork.voip.ms

session transport udp

dtmf-relay rtp-nte

below is my sip trace. Specific inforation removed and replaced with X's

*Mar 12 17:37:40.471: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

INVITE sip:XXX9177522@50.20.134.24:5060 SIP/2.0

Via: SIP/2.0/UDP 74.63.41.218:5060;branch=z9hG4bK05a308ce

Max-Forwards: 70

From: "GARRETT " ;tag=as0009de80

To:

Contact:

Call-ID: 0468fe2415fdcd47074de44c25f3f433@74.63.41.218:5060

CSeq: 102 INVITE

User-Agent: voip.ms

Date: Tue, 12 Mar 2013 16:54:32 GMT

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

Supported: replaces, timer

Remote-Party-ID: "GARRETT " ;party=calling;privacy=off;screen=no

Content-Type: application/sdp

Content-Length: 268

v=0

o=root 805135302 805135302 IN IP4 74.63.41.218

s=voip.ms

c=IN IP4 74.63.41.218

t=0 0

m=audio 16470 RTP/AVP 0 18 101

a=rtpmap:0 PCMU/8000

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=sendrecv

*Mar 12 17:37:40.487: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 74.63.41.218:5060;branch=z9hG4bK05a308ce

From: "GARRETT " ;tag=as0009de80

To:

Date: Tue, 12 Mar 2013 17:37:40 GMT

Call-ID: 0468fe2415fdcd47074de44c25f3f433@74.63.41.218:5060

CSeq: 102 INVITE

Allow-Events: telephone-event

Server: Cisco-SIPGateway/IOS-12.x

Content-Length: 0

*Mar 12 17:37:40.487: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

INVITE sip:XXX9177522@192.168.10.51:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.10.52:5060;branch=z9hG4bK6B2285

Remote-Party-ID: "GARRETT " ;party=calling;screen=no;privacy=off

From: "GARRETT " <>XXXXXX1293@newyork.voip.ms>;tag=3AC8B58-E0D

To:

Date: Tue, 12 Mar 2013 17:37:40 GMT

Call-ID: 5F5575A3-8A7211E2-831CEADC-2EFD56DE@192.168.10.52

Supported: 100rel,timer,resource-priority,replaces

Min-SE: 1800

Cisco-Guid: 1599316115-2322731490-2199317212-788354782

User-Agent: Cisco-SIPGateway/IOS-12.x

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

CSeq: 101 INVITE

Timestamp: 1363109860

Contact:

Expires: 180

Allow-Events: telephone-event

Max-Forwards: 69

Content-Type: application/sdp

Content-Disposition: session;handling=required

Content-Length: 297

v=0

o=CiscoSystemsSIP-GW-UserAgent 6223 2315 IN IP4 192.168.10.52

s=SIP Call

c=IN IP4 192.168.10.52

t=0 0

m=audio 19064 RTP/AVP 18 101 19

c=IN IP4 192.168.10.52

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=rtpmap:19 CN/8000

a=ptime:20

*Mar 12 17:37:40.495: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 192.168.10.52:5060;branch=z9hG4bK6B2285

From: "GARRETT " <>XXXXXX1293@newyork.voip.ms>;tag=3AC8B58-E0D

To:

Date: Tue, 12 Mar 2013 16:54:32 GMT

Call-ID: 5F5575A3-8A7211E2-831CEADC-2EFD56DE@192.168.10.52

CSeq: 101 INVITE

Allow-Events: presence

Content-Length: 0

*Mar 12 17:37:40.499: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

INVITE sip:17175193304@192.168.10.52:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.10.51:5060;branch=z9hG4bK122b41b544

From: "GARRETT " ;tag=a423d606-f845-422c-bf66-e67421ed9f12-33127376

To: <17175193304>

Date: Tue, 12 Mar 2013 16:54:32 GMT

Call-ID: 81ab5980-13f15dc8-13-330aa8c0@192.168.10.51

Supported: timer,resource-priority,replaces

Min-SE: 1800

User-Agent: Cisco-CUCM8.0

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

CSeq: 101 INVITE

Contact:

Expires: 180

Allow-Events: presence

Supported: X-cisco-srtp-fallback

Supported: Geolocation

Cisco-Guid: 2175490432-0000065536-0000000019-0856336576

Session-Expires: 1800

P-Asserted-Identity: "GARRETT "

Remote-Party-ID: "GARRETT " ;party=calling;screen=yes;privacy=off

Max-Forwards: 68

Content-Length: 0

*Mar 12 17:37:40.511: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 192.168.10.51:5060;branch=z9hG4bK122b41b544

From: "GARRETT " ;tag=a423d606-f845-422c-bf66-e67421ed9f12-33127376

To: <17175193304>

Date: Tue, 12 Mar 2013 17:37:40 GMT

Call-ID: 81ab5980-13f15dc8-13-330aa8c0@192.168.10.51

CSeq: 101 INVITE

Allow-Events: telephone-event

Server: Cisco-SIPGateway/IOS-12.x

Content-Length: 0

*Mar 12 17:37:40.511: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 404 Not Found

Via: SIP/2.0/UDP 192.168.10.51:5060;branch=z9hG4bK122b41b544

From: "GARRETT " ;tag=a423d606-f845-422c-bf66-e67421ed9f12-33127376

To: <17175193304>;tag=3AC8B70-218F

Date: Tue, 12 Mar 2013 17:37:40 GMT

Call-ID: 81ab5980-13f15dc8-13-330aa8c0@192.168.10.51

CSeq: 101 INVITE

Allow-Events: telephone-event

Server: Cisco-SIPGateway/IOS-12.x

Reason: Q.850;cause=1

Content-Length: 0

*Mar 12 17:37:40.515: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

ACK sip:17175193304@192.168.10.52:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.10.51:5060;branch=z9hG4bK122b41b544

From: "GARRETT " ;tag=a423d606-f845-422c-bf66-e67421ed9f12-33127376

To: <17175193304>;tag=3AC8B70-218F

Date: Tue, 12 Mar 2013 16:54:32 GMT

Call-ID: 81ab5980-13f15dc8-13-330aa8c0@192.168.10.51

Max-Forwards: 70

CSeq: 101 ACK

Allow-Events: presence

Content-Length: 0

*Mar 12 17:37:40.519: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 404 Not Found

Via: SIP/2.0/UDP 192.168.10.52:5060;branch=z9hG4bK6B2285

From: "GARRETT " <>XXXXXX1293@newyork.voip.ms>;tag=3AC8B58-E0D

To: ;tag=a423d606-f845-422c-bf66-e67421ed9f12-33127373

Date: Tue, 12 Mar 2013 16:54:32 GMT

Call-ID: 5F5575A3-8A7211E2-831CEADC-2EFD56DE@192.168.10.52

CSeq: 101 INVITE

Allow-Events: presence

Reason: Q.850;cause=1

Content-Length: 0

*Mar 12 17:37:40.523: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

ACK sip:XXX9177522@192.168.10.51:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.10.52:5060;branch=z9hG4bK6B2285

From: "GARRETT " <>XXXXXX1293@newyork.voip.ms>;tag=3AC8B58-E0D

To: ;tag=a423d606-f845-422c-bf66-e67421ed9f12-33127373

Date: Tue, 12 Mar 2013 17:37:40 GMT

Call-ID: 5F5575A3-8A7211E2-831CEADC-2EFD56DE@192.168.10.52

Max-Forwards: 70

CSeq: 101 ACK

Allow-Events: telephone-event

Content-Length: 0

*Mar 12 17:37:40.527: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 404 Not Found

Via: SIP/2.0/UDP 74.63.41.218:5060;branch=z9hG4bK05a308ce

From: "GARRETT " ;tag=as0009de80

To: ;tag=3AC8B7C-1E31

Date: Tue, 12 Mar 2013 17:37:40 GMT

Call-ID: 0468fe2415fdcd47074de44c25f3f433@74.63.41.218:5060

CSeq: 102 INVITE

Allow-Events: telephone-event

Server: Cisco-SIPGateway/IOS-12.x

Reason: Q.850;cause=1

Content-Length: 0

*Mar 12 17:37:40.611: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

ACK sip:XXX9177522@50.20.134.24:5060 SIP/2.0

Via: SIP/2.0/UDP 74.63.41.218:5060;branch=z9hG4bK05a308ce

Max-Forwards: 70

From: "GARRETT " ;tag=as0009de80

To: ;tag=3AC8B7C-1E31

Contact:

Call-ID: 0468fe2415fdcd47074de44c25f3f433@74.63.41.218:5060

CSeq: 102 ACK

User-Agent: voip.ms

Content-Length: 0

rbblue234 · ‎03-12-2013

i have this solved but not how i wanted too.

I went into my VoIP proivder and turned off the authenication for inound calls. Since the GW will have a static, i told my VoIP proivder the IP and it will deliver every call too that IP.

every time i entered the below commands... my inbound and outbound calls failed

config t

voice service voip

sip

outbound-proxy dns:newyork.voip.ms

Ayodeji Okanlawon · ‎03-13-2013

Oh I can see where the error is..The destination-pattern is pointing back to CUCM. It should point to your sip provider

Try this..

Configure inbound dial-peer as follows

dial-peer voice 101 voip

voice-class sip options-ping 60

session protocol sipv2

session transport udp

incoming called-number . ( use . here)

dtmf-relay rtp-nte

Configure outbound dial-peer as follows

dial-peer voice 100 voip

destination-pattern XXXT ( this will match all the numbers you want to send out, change this to meet your dialled numbers)

voice-class sip options-ping 60

session protocol sipv2

session target ipv4:YYYYYYY (yyy=sip provider ip address

session transport udp

dtmf-relay rtp-nte

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