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New Member

Problems with phone communication (SIP & SCCP)

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?

9 REPLIES
Hall of Fame Super Gold

Problems with phone communication (SIP & SCCP)

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

New Member

Problems with phone communication (SIP & SCCP)

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.

New Member

Problems with phone communication (SIP & SCCP)

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

Hall of Fame Super Gold

Problems with phone communication (SIP & SCCP)

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

New Member

Problems with phone communication (SIP & SCCP)

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

Hall of Fame Super Gold

Problems with phone communication (SIP & SCCP)

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.

New Member

Problems with phone communication (SIP & SCCP)

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.

Hall of Fame Super Gold

Problems with phone communication (SIP & SCCP)

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.

New Member

Problems with phone communication (SIP & SCCP)

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:

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