Problems with phone communication (SIP & SCCP)

[Pawel Lenart]

Hi, On my current system I'm running some tests using 2 x 9971 (SIP) + 1 x 7965 on 2921 router. I can make outgoing phonecalls from any phone, I can receive phonecalls on 7965 (from any 9971 phone) but I can't make calls between 2 9971 units and from 7965 to 9971. Any idea why I have this problem?

[paolo bevilacqua]

Likely something is configured wrong, but we can't tell what, because you have not included you relevant configuration.

[Pawel Lenart]

Sorry about this. I though someone had this problem and may know the answer straight away. My config:

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

  bind control source-interface GigabitEthernet0/1

  bind media source-interface GigabitEthernet0/2

  registrar server expires max 3600 min 3600

  localhost dns: SIP PROVIDER

  outbound-proxy dns:SIP PROVIDER

  no update-callerid

  sip-profiles 1000

!

voice class codec 1

codec preference 1 g711ulaw

codec preference 2 g711alaw

codec preference 3 g728

codec preference 4 g726r16

video codec h264

!

voice class sip-profiles 1000

request ANY sdp-header Connection-Info remove

response ANY sdp-header Connection-Info remove

request ANY sdp-header Connection-Info remove

response ANY sdp-header Connection-Info remove

!

voice class sip-profiles 100

!

!

voice register global

mode cme

source-address ROUTER IP ADDRESS port 5060

max-dn 5

max-pool 20

load 9971 sip9971.9-1-1SR1

authenticate register

authenticate realm SIP PROVIDER

timezone 21

time-format 24

date-format D/M/Y

tftp-path flash:

create profile sync 0730901611676053

network-locale GB

ntp-server ROUTER IP ADDRESS mode directedbroadcast

camera

video

!

voice register dn  1

number 100

name FIRSTNAME LASTNAME

label FL - 100

!

voice register dn  2

number 101

name FIRSTNAME1 LASTNAME1

label F1L1 - 101

!

voice register pool  1

id mac PHONE MAC ADDRESS

type 9971

number 1 dn 1

voice-class codec 1

username FL password 123456

speed-dial 1 90111111111 label "COMPANY"

camera

video

!

voice register pool  2

id mac PHONE MAC ADDRESS

type 9971

number 1 dn 2

voice-class codec 1

username F1L1 password 123456

camera

video

dial-peer voice 1 voip

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

session protocol sipv2

session target sip-server

incoming called-number .T

dtmf-relay rtp-nte

!

dial-peer voice 1000 voip

description ** Incoming call from SIP trunk (Generic SIP Trunk Provider) **

translation-profile incoming INCOMING

translation-profile outgoing OUTGOING

session protocol sipv2

session target sip-server

incoming called-number SIP PROVIDER

voice-class codec 1

dtmf-relay rtp-nte

ip qos dscp cs5 media

ip qos dscp cs4 signaling

no vad

!

dial-peer voice 1001 voip

description ** star code to SIP trunk (Generic SIP Trunk Provider) **

destination-pattern *..

session protocol sipv2

session target sip-server

voice-class codec 1

voice-class sip dtmf-relay force rtp-nte

dtmf-relay rtp-nte

ip qos dscp cs5 media

ip qos dscp cs4 signaling

no vad

!

dial-peer voice 2000 voip

description OUTBOUND

translation-profile incoming SIP_Incoming

translation-profile outgoing OUTGOING

destination-pattern 9T

session protocol sipv2

session target sip-server

voice-class codec 1

dtmf-relay rtp-nte

no vad

!

!

sip-ua

credentials username SIP PROVIDER password 7 SIP PROVIDER realm SIP PROVIDER

authentication username SIP PROVIDER password 7 SIP PROVIDER

no remote-party-id

retry invite 2

retry register 10

timers connect 100

registrar dns:SIP PROVIDER expires 3600

sip-server dns:SIP PROVIDER

host-registrar

!

!

!

gatekeeper

shutdown

!

!

telephony-service

no auto-reg-ephone

pin 123456 override

max-ephones 20

max-dn 100

ip source-address ROUTER IP ADDRESS port 2000

system message COMPANY NAME

cnf-file location flash:

load 7945 SCCP45.9-1-1SR1S

load 7965 SCCP45.9-1-1SR1S.loads

load 7970 SCCP41.9-1-1SR1S

load 7971 SCCP70.9-1-1SR1S.loads

time-format 24

date-format dd-mm-yy

max-conferences 8 gain -6

web admin system name USERNAME secret 5 PASSWORD

transfer-system full-consult

directory entry 1 9011111111 name COMPANY

directory entry 2 0111111 name COMPANYNO9

create cnf-files version-stamp 7960 Jan 13 2012 12:38:29

!

!

!

ephone-dn  2  dual-line

number 202 secondary SIP PROVIDER no-reg both

label 202

description FIRSTNAME2 LASTNAME2

name FIRSTNAME2 LASTNAME2

!

!

ephone  1

mac-address PHONE MAC ADDRESS

username "F2L2" password 123456

speed-dial 1 90111111# label "COMPANY"

type 7945

button  1:2

pin 123456

If you need anything else please let me know.

[Pawel Lenart]

I'm guessing this problem is harder then I expected. Anybody?

[paolo bevilacqua]

Take "debug ccsip message" with "term mon" for a failed call.

[Pawel Lenart]

Output:

INVITE sip:101@ROUTER IP ADDRESS SIP/2.0

Via: SIP/2.0/UDP PHONE IP ADDRESS:5060;branch=z9hG4bK168a26c1

From: "FIRSTNAME LASTNAME" <100>;tag=ecc882b1480f003416165545-6db188e4

To: <101>

Call-ID: ecc882b1-480f000f-07610812-2058f5c3@PHONE IP ADDRESS

Max-Forwards: 70

Date: Tue, 17 Jan 2012 12:25:21 GMT

CSeq: 101 INVITE

User-Agent: Cisco-CP9971/9.1.1

Contact: <100>

Expires: 180

Accept: application/sdp

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

Remote-Party-ID: "FIRSTNAME LASTNAME" <100>;party=calling;id-type=subscriber;privacy=off;screen=yes

Supported: replaces,join,sdp-anat,norefersub,extended-refer,X-cisco-callinfo,X-cisco-serviceuri,X-cisco-escapecodes,X-cisco-service-control,X-cisco-srtp-fallback,X-cisco-monrec,X-cisco-config,X-cisco-sis-5.1.0,X-cisco-xsi-8.0.1

Allow-Events: kpml,dialog

Content-Length: 566

Content-Type: application/sdp

Content-Disposition: session;handling=optional

v=0

o=Cisco-SIPUA 4580 0 IN IP4 PHONE IP ADDRESS

s=SIP Call

t=0 0

m=audio 23686 RTP/AVP 0 8 18 102 116 124 101

c=IN IP4 PHONE IP ADDRESS

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:102 L16/16000

a=rtpmap:116 iLBC/8000

a=fmtp:116 mode=20

a=rtpmap:124 ISAC/16000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=sendrecv

m=video 24402 RTP/AVP 97

c=IN IP4 PHONE IP ADDRESS

b=TIAS:1000000

a=rtpmap:97 H264/90000

a=fmtp:97 profile-level-id=42801E;packetization-mode=0;level-asymmetry-allowed=1

a=sendrecv

*Jan 17 12:25:29.672: //274/2D6637CF80D0/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP PHONE IP ADDRESS:5060;branch=z9hG4bK168a26c1

From: "FIRSTNAME LASTNAME" <100>;tag=ecc882b1480f003416165545-6db188e4

To: <101>

Date: Tue, 17 Jan 2012 12:25:29 GMT

Call-ID: ecc882b1-480f000f-07610812-2058f5c3@PHONE IP ADDRESS

CSeq: 101 INVITE

Allow-Events: telephone-event

Server: Cisco-SIPGateway/IOS-12.x

Content-Length: 0

*Jan 17 12:25:29.680: //276/2D6637CF80D0/SIP/Msg/ccsipDisplayMsg:

Sent:

INVITE sip:101@PHONE IP ADDRESS:5060 SIP/2.0

Via: SIP/2.0/UDP ROUTER IP ADDRESS:5060;branch=z9hG4bKEA500

From: "FIRSTNAME LASTNAME" <100>;tag=679614-166B

To: <101>

Date: Tue, 17 Jan 2012 12:25:29 GMT

Call-ID: 2D6770B8-403D11E1-80D68BB4-5332D2E3@SIP PROVIDER

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

Min-SE:  1800

Cisco-Guid: 0761673679-1077744097-2161150900-1395839715

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

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

CSeq: 101 INVITE

Timestamp: 1326803129

Contact: <100>

Expires: 180

Allow-Events: telephone-event

Max-Forwards: 69

Content-Type: application/sdp

Content-Disposition: session;handling=required

Content-Length: 357

v=0

o=CiscoSystemsSIP-GW-UserAgent 3239 9064 IN IP4 ROUTER IP ADDRESS

s=SIP Call

t=0 0

m=audio 29358 RTP/AVP 0 8 19

c=IN IP4 PUBLIC IP ADDRESS

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:19 CN/8000

m=video 24378 RTP/AVP 119

c=IN IP4 PUBLIC IP ADDRESS

b=TIAS:1000000

a=rtpmap:119 H264/90000

a=fmtp:119 profile-level-id=42801E;packetization-mode=0

*Jan 17 12:25:29.732: //276/2D6637CF80D0/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 100 Giving a try

Via: SIP/2.0/UDP ROUTER IP ADDRESS:5060;branch=z9hG4bKEA500;rport=61112;received=80.229.165.140

From: "FIRSTNAME LASTNAME" <100>;tag=679614-166B

To: <101>

Call-ID: 2D6770B8-403D11E1-80D68BB4-5332D2E3@SIP PROVIDER

CSeq: 101 INVITE

Server: OpenSIPS (1.6.2-notls (i386/linux))

Content-Length: 0

*Jan 17 12:25:29.732: //276/2D6637CF80D0/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 407 Proxy Authentication Required

Via: SIP/2.0/UDP ROUTER IP ADDRESS:5060;rport=61112;received=80.229.165.140;branch=z9hG4bKEA500

From: "FIRSTNAME LASTNAME" <100>;tag=679614-166B

To: <101>;tag=fd79486175647ed1617969929fdaad02.5d24

Call-ID: 2D6770B8-403D11E1-80D68BB4-5332D2E3@SIP PROVIDER

CSeq: 101 INVITE

Proxy-Authenticate: Digest realm="SIP PROVIDER", nonce="4f1568d70000420297873fc5168fe3b618b379f3e11811af"

Server: OpenSIPS (1.5.3-notls (x86_64/linux))

Content-Length: 0

*Jan 17 12:25:29.732: //276/2D6637CF80D0/SIP/Msg/ccsipDisplayMsg:

Sent:

ACK sip:101@PHONE IP ADDRESS:5060 SIP/2.0

Via: SIP/2.0/UDP ROUTER IP ADDRESS:5060;branch=z9hG4bKEA500

From: "FIRSTNAME LASTNAME" <100>;tag=679614-166B

To: <101>;tag=fd79486175647ed1617969929fdaad02.5d24

Date: Tue, 17 Jan 2012 12:25:29 GMT

Call-ID: 2D6770B8-403D11E1-80D68BB4-5332D2E3@SIP PROVIDER

Max-Forwards: 70

CSeq: 101 ACK

Allow-Events: telephone-event

Content-Length: 0

*Jan 17 12:25:29.732: //276/2D6637CF80D0/SIP/Msg/ccsipDisplayMsg:

Sent:

INVITE sip:101@PHONE IP ADDRESS:5060 SIP/2.0

Via: SIP/2.0/UDP ROUTER IP ADDRESS:5060;branch=z9hG4bKEB1EBD

From: "FIRSTNAME LASTNAME" <100>;tag=679614-166B

To: <101>

Date: Tue, 17 Jan 2012 12:25:29 GMT

Call-ID: 2D6770B8-403D11E1-80D68BB4-5332D2E3@SIP PROVIDER

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

Min-SE:  1800

Cisco-Guid: 0761673679-1077744097-2161150900-1395839715

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

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

CSeq: 102 INVITE

Timestamp: 1326803129

Contact: <100>

Expires: 180

Allow-Events: telephone-event

Proxy-Authorization: Digest username="SIP USERNAME",realm="SIP PROVIDER",uri="sip:101@PHONE IP ADDRESS:5060",response="a4037cb65d034dc4850363bbb6489201",nonce="4f1568d70000420297873fc5168fe3b618b379f3e11811af",algorithm=md5

Max-Forwards: 69

Content-Type: application/sdp

Content-Disposition: session;handling=required

Content-Length: 357

v=0

o=CiscoSystemsSIP-GW-UserAgent 3239 9064 IN IP4 ROUTER IP ADDRESS

s=SIP Call

t=0 0

m=audio 29358 RTP/AVP 0 8 19

c=IN IP4 PUBLIC IP ADDRESS

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:19 CN/8000

m=video 24378 RTP/AVP 119

c=IN IP4 PUBLIC IP ADDRESS

b=TIAS:1000000

a=rtpmap:119 H264/90000

a=fmtp:119 profile-level-id=42801E;packetization-mode=0

*Jan 17 12:25:29.792: //276/2D6637CF80D0/SIP/Msg/ccsipDisplayMsg:

Received:

R1003326#SIP/2.0 100 Giving a try

Via: SIP/2.0/UDP ROUTER IP ADDRESS:5060;branch=z9hG4bKEB1EBD;rport=61112;received=80.229.165.140

From: "FIRSTNAME LASTNAME" <100>;tag=679614-166B

To: <101>

Call-ID: 2D6770B8-403D11E1-80D68BB4-5332D2E3@SIP PROVIDER

CSeq: 102 INVITE

Server: OpenSIPS (1.6.2-notls (i386/linux))

Content-Length: 0

R1003326#

*Jan 17 12:25:34.924: //276/2D6637CF80D0/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 408 Request Timeout

Via: SIP/2.0/UDP ROUTER IP ADDRESS:5060;rport=61112;received=80.229.165.140;branch=z9hG4bKEB1EBD

From: "FIRSTNAME LASTNAME" <100>;tag=679614-166B

To: <101>;tag=c4734c5d85c7ed191290cd06ad14c355-5855

Call-ID: 2D6770B8-403D11E1-80D68BB4-5332D2E3@SIP PROVIDER

CSeq: 102 INVITE

Server: OpenSIPS (1.5.3-notls (x86_64/linux))

Content-Length: 0

*Jan 17 12:25:34.924: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

ACK sip:101@PHONE IP ADDRESS:5060 SIP/2.0

Via: SIP/2.0/UDP ROUTER IP ADDRESS:5060;branch=z9hG4bKEB1EBD

From: "FIRSTNAME LASTNAME" <100>;tag=679614-166B

To: <101>;tag=c4734c5d85c7ed191290cd06ad14c355-5855

Date: Tue, 17 Jan 2012 12:25:29 GMT

Call-ID: 2D6770B8-403D11E1-80D68BB4-5332D2E3@SIP PROVIDER

Max-Forwards: 70

CSeq: 102 ACK

Allow-Events: telephone-event

Content-Length: 0

*Jan 17 12:25:34.928: //277/2D6637CF80D0/SIP/Msg/ccsipDisplayMsg:

Sent:

INVITE sip:101@PHONE 2 IP ADDRESS:5060 SIP/2.0

Via: SIP/2.0/UDP ROUTER IP ADDRESS:5060;branch=z9hG4bKEC185C

From: "FIRSTNAME LASTNAME" <100>;tag=67AA98-1E10

To: <101>

Date: Tue, 17 Jan 2012 12:25:34 GMT

Call-ID: 30883974-403D11E1-80D88BB4-5332D2E3@SIP PROVIDER

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

Min-SE:  1800

Cisco-Guid: 0761673679-1077744097-2161150900-1395839715

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

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

CSeq: 101 INVITE

Timestamp: 1326803134

Contact: <100>

Expires: 180

Allow-Events: telephone-event

Max-Forwards: 69

Content-Type: application/sdp

Content-Disposition: session;handling=required

Content-Length: 357

v=0

o=CiscoSystemsSIP-GW-UserAgent 9930 5858 IN IP4 ROUTER IP ADDRESS

s=SIP Call

t=0 0

m=audio 19190 RTP/AVP 0 8 19

c=IN IP4 PUBLIC IP ADDRESS

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:19 CN/8000

m=video 32648 RTP/AVP 119

c=IN IP4 PUBLIC IP ADDRESS

b=TIAS:1000000

a=rtpmap:119 H264/90000

a=fmtp:119 profile-level-id=42801E;packetization-mode=0

*Jan 17 12:25:34.984: //277/2D6637CF80D0/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 100 Giving a try

Via: SIP/2.0/UDP ROUTER IP ADDRESS:5060;branch=z9hG4bKEC185C;rport=61112;received=80.229.165.140

From: "FIRSTNAME LASTNAME" <100>;tag=67AA98-1E10

To: <101>

Call-ID: 30883974-403D11E1-80D88BB4-5332D2E3@SIP PROVIDER

CSeq: 101 INVITE

Server: OpenSIPS (1.6.2-notls (i386/linux))

Content-Length: 0

*Jan 17 12:25:35.056: //277/2D6637CF80D0/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 407 Proxy Authentication Required

Via: SIP/2.0/UDP ROUTER IP ADDRESS:5060;rport=61112;received=80.229.165.140;branch=z9hG4bKEC185C

From: "FIRSTNAME LASTNAME" <100>;tag=67AA98-1E10

To: <101>;tag=fd79486175647ed1617969929fdaad02.3d64

Call-ID: 30883974-403D11E1-80D88BB4-5332D2E3@SIP PROVIDER

CSeq: 101 INVITE

Proxy-Authenticate: Digest realm="SIP PROVIDER", nonce="4f1568dc0000443b8f9419759275ea5498a472dce099c8eb"

Server: OpenSIPS (1.5.3-notls (x86_64/linux))

Content-Length: 0

*Jan 17 12:25:35.060: //277/2D6637CF80D0/SIP/Msg/ccsipDisplayMsg:

Sent:

ACK sip:101@PHONE 2 IP ADDRESS:5060 SIP/2.0

Via: SIP/2.0/UDP ROUTER IP ADDRESS:5060;branch=z9hG4bKEC185C

From: "FIRSTNAME LASTNAME" <100>;tag=67AA98-1E10

To: <101>;tag=fd79486175647ed1617969929fdaad02.3d64

Date: Tue, 17 Jan 2012 12:25:34 GMT

Call-ID: 30883974-403D11E1-80D88BB4-5332D2E3@SIP PROVIDER

Max-Forwards: 70

CSeq: 101 ACK

Allow-Events: telephone-event

Content-Length: 0

*Jan 17 12:25:35.060: //277/2D6637CF80D0/SIP/Msg/ccsipDisplayMsg:

Sent:

INVITE sip:101@PHONE 2 IP ADDRESS:5060 SIP/2.0

Via: SIP/2.0/UDP ROUTER IP ADDRESS:5060;branch=z9hG4bKED1288

From: "FIRSTNAME LASTNAME" <100>;tag=67AA98-1E10

To: <101>

Date: Tue, 17 Jan 2012 12:25:35 GMT

Call-ID: 30883974-403D11E1-80D88BB4-5332D2E3@SIP PROVIDER

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

Min-SE:  1800

Cisco-Guid: 0761673679-1077744097-2161150900-1395839715

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

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

CSeq: 102 INVITE

Timestamp: 1326803135

Contact: <100>

Expires: 180

Allow-Events: telephone-event

Proxy-Authorization: Digest username="SIP USERNAME",realm="SIP PROVIDER",uri="sip:101@PHONE 2 IP ADDRESS:5060",response="3e71129eb31c679e16db0b290f54e512",nonce="4f1568dc0000443b8f9419759275ea5498a472dce099c8eb",algorithm=md5

Max-Forwards: 69

Content-Type: application/sdp

Content-Disposition: session;handling=required

Content-Length: 357

v=0

o=CiscoSystemsSIP-GW-UserAgent 9930 5858 IN IP4 ROUTER IP ADDRESS

s=SIP Call

t=0 0

m=audio 19190 RTP/AVP 0 8 19

c=IN IP4 PUBLIC IP ADDRESS

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:19 CN/8000

m=video 32648 RTP/AVP 119

c=IN IP4 PUBLIC IP ADDRESS

b=TIAS:1000000

a=rtpmap:119 H264/90000

a=fmtp:119 profile-level-id=42801E;packetization-mode=0

*Jan 17 12:25:35.124: //277/2D6637CF80D0/SIP/Msg/ccsipDisplayMsg:

Received:

R1003326#SIP/2.0 100 Giving a try

Via: SIP/2.0/UDP ROUTER IP ADDRESS:5060;branch=z9hG4bKED1288;rport=61112;received=80.229.165.140

From: "FIRSTNAME LASTNAME" <100>;tag=67AA98-1E10

To: <101>

Call-ID: 30883974-403D11E1-80D88BB4-5332D2E3@SIP PROVIDER

CSeq: 102 INVITE

Server: OpenSIPS (1.6.2-notls (i386/linux))

Content-Length: 0

*Jan 17 12:25:36.060: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

...

*Jan 17 12:25:36.064: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

...

R1003326#

*Jan 17 12:25:40.308: //277/2D6637CF80D0/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 408 Request Timeout

Via: SIP/2.0/UDP ROUTER IP ADDRESS:5060;rport=61112;received=80.229.165.140;branch=z9hG4bKED1288

From: "FIRSTNAME LASTNAME" <100>;tag=67AA98-1E10

To: <101>;tag=c4734c5d85c7ed191290cd06ad14c355-9942

Call-ID: 30883974-403D11E1-80D88BB4-5332D2E3@SIP PROVIDER

CSeq: 102 INVITE

Server: OpenSIPS (1.5.3-notls (x86_64/linux))

Content-Length: 0

*Jan 17 12:25:40.312: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

ACK sip:101@PHONE 2 IP ADDRESS:5060 SIP/2.0

Via: SIP/2.0/UDP ROUTER IP ADDRESS:5060;branch=z9hG4bKED1288

From: "FIRSTNAME LASTNAME" <100>;tag=67AA98-1E10

To: <101>;tag=c4734c5d85c7ed191290cd06ad14c355-9942

Date: Tue, 17 Jan 2012 12:25:35 GMT

Call-ID: 30883974-403D11E1-80D88BB4-5332D2E3@SIP PROVIDER

Max-Forwards: 70

CSeq: 102 ACK

Allow-Events: telephone-event

Content-Length: 0

*Jan 17 12:25:40.312: //274/2D6637CF80D0/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 408 Request Timeout

Via: SIP/2.0/UDP PHONE IP ADDRESS:5060;branch=z9hG4bK168a26c1

From: "FIRSTNAME LASTNAME" <100>;tag=ecc882b1480f003416165545-6db188e4

To: <101>;tag=67BFA0-930

Date: Tue, 17 Jan 2012 12:25:29 GMT

Call-ID: ecc882b1-480f000f-07610812-2058f5c3@PHONE IP ADDRESS

CSeq: 101 INVITE

Allow-Events: telephone-event

Server: Cisco-SIPGateway/IOS-12.x

Reason: Q.850;cause=102

Content-Length: 0

*Jan 17 12:25:40.316: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

R1003326#ACK sip:101@ROUTER IP ADDRESS SIP/2.0

Via: SIP/2.0/UDP PHONE IP ADDRESS:5060;branch=z9hG4bK168a26c1

From: "FIRSTNAME LASTNAME" <100>;tag=ecc882b1480f003416165545-6db188e4

To: <101>;tag=67BFA0-930

Call-ID: ecc882b1-480f000f-07610812-2058f5c3@PHONE IP ADDRESS

Date: Tue, 17 Jan 2012 12:25:32 GMT

CSeq: 101 ACK

Content-Length: 0

*Jan 17 12:25:42.156: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

OPTIONS sip:SIP USERNAME@ROUTER IP ADDRESS:5060 SIP/2.0

Via: SIP/2.0/UDP SIP TRUNK PROVIDER IP ADDRESS:5060;branch=0

From: sip:vnt-10@SIP PROVIDER;tag=47a92cf4

To: sip:SIP USERNAME@SIP PROVIDER

Call-ID: b32b89b2-23251c16-61877@SIP TRUNK PROVIDER IP ADDRESS

CSeq: 1 OPTIONS

Content-Length: 0

*Jan 17 12:25:42.160: //278/34D7C09580D9/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 200 OK

Via: SIP/2.0/UDP SIP TRUNK PROVIDER IP ADDRESS:5060;branch=0

From: sip:vnt-10@SIP PROVIDER;tag=47a92cf4

To: sip:SIP USERNAME@SIP PROVIDER;tag=67C6D4-1E5B

Date: Tue, 17 Jan 2012 12:25:42 GMT

Call-ID: b32b89b2-23251c16-61877@SIP TRUNK PROVIDER IP ADDRESS

Server: Cisco-SIPGateway/IOS-12.x

CSeq: 1 OPTIONS

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

Allow-Events: telephone-event

Accept: application/sdp

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

Conte

R1003326#nt-Type: application/sdp

Content-Length: 148

v=0

o=CiscoSystemsSIP-GW-UserAgent 3953 4331 IN IP4 ROUTER IP ADDRESS

s=SIP Call

t=0 0

m=audio 0 RTP/AVP 18 0 8 9 4 2 15

c=IN IP4 PUBLIC IP ADDRESS

[paolo bevilacqua]

Before, you mentioned about failure to call a SIP phones, never mentioned about SIP trunk.

Check that the SIP phone is registered, or include a trace again.

[Pawel Lenart]

SIP trunk is working fine. I can make phonecalls from any phone and at the moment I have incomming phonecalls to go to SCCP phone and they are working fine as well. Problem is with phoning in to SIP phone.

[paolo bevilacqua]

The trace you posted begins with INVITE from router to phone.

It should begin with INVITE from ITSP to phone.

Please take the trace again correctly.

[Pawel Lenart]

Hi Paolo,

is this waht you need:

Sent:

REGISTER sip:SIP TRUNK PROVIDER:5060 SIP/2.0

Via: SIP/2.0/UDP ROUTER IP ADDRESS:5060;branch=z9hG4bKC4A52E

From: <103>;tag=5683BE4-15D7

To: <103>

Date: Wed, 18 Jan 2012 11:44:11 GMT

Call-ID: FE0B4FED-404D11E1-825C8BB4-5332D2E3

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

Max-Forwards: 70

Timestamp: 1326887051

CSeq: 854 REGISTER

Contact: <103>

Expires:  3600

Supported: path

Content-Length: 0

Jan 18 11:44:11.902: //4621/000000000000/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 401 Unauthorized

Via: SIP/2.0/UDP ROUTER IP ADDRESS:5060;rport=61112;received=SIP TRUNK PROVIDER;branch=z9hG4bKC4A52E

From: <103>;tag=5683BE4-15D7

To: <103>;tag=fd79486175647ed1617969929fdaad02.706d

Call-ID: FE0B4FED-404D11E1-825C8BB4-5332D2E3

CSeq: 854 REGISTER

WWW-Authenticate: Digest realm="SIP TRUNK PROVIDER", nonce="4f16b0b00000a3878d01a78a977ca08aa44be32b112089ea"

Server: OpenSIPS (1.5.3-notls (x86_64/linux))

Content-Length: 0

Jan 18 11:44:11.906: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: