CME - inbound and outbound call problem

zittastanislav · ‎11-24-2012

Greetings,

i am trying to configure CME. It sits on perimeter between internal network where are SCCP and SIP phones. Internal calls are working properly but Call from outside to inside is impossible. Sip messages are saying that internal phone is ringing but i cannot hear anything. Beside that, i'm not in place of cme so i cannot hear if phone is actually ringing, but in my cell phone, i hear no ringback. Outside calls aren't working too.

I am connected through SIP trunk to outside world. My ITSP told me, that no registration is needed aganist their technology.

Here is my config:

service password-encryption

!

hostname voip-router

!

boot-start-marker

boot-end-marker

!

enable secret 4 z06kVgpORxSKsT1dQ0iSEz0i8V605qY/bmx7CWxMnLE

!

no aaa new-model

!

dot11 syslog

ip source-route

!

ip cef

!

ip dhcp binding cleanup interval 300

ip dhcp excluded-address XXXX XXXX

ip dhcp excluded-address XXXX

!

ip domain name XXXXXXXXXXXXXXXXXXXXX

ip name-server XXXX

ip name-server XXXXX

no ipv6 cef

!

multilink bundle-name authenticated

!

voice rtp send-recv

!

voice service voip

ip address trusted list

ipv4 XXXXX

no ip address trusted authenticate

allow-connections h323 to sip

allow-connections sip to h323

allow-connections sip to sip

sip

registrar server

!

voice register global

mode cme

source-address XXXX port 5060

max-dn 192

max-pool 58

authenticate register

create profile sync 0001227300540304

!

voice-card 0

!

license ....

!

redundancy

!

translation-rule 1

Rule 1 202 XXXXXXXXX

!

interface GigabitEthernet0/0

ip address...

duplex auto

speed auto

!

interface GigabitEthernet0/1

ip address ...

duplex auto

speed auto

!

ip forward-protocol nd

ip http server

ip http authentication local

ip http secure-server

!

!

tftp-server flash:spa50x_30x_cz_v753.xml alias spa50x_30x_cz_v753.xml

tftp-server flash:/cs-be-sccp.jar alias user_define_1/be-sccp.jar

tftp-server flash:/649EF376D356.xml alias 210.xml

tftp-server flash:211.xml

!

control-plane

!

call threshold global cpu-avg low 68 high 75

call threshold global total-mem low 75 high 85

!

mgcp profile default

!

dial-peer voice 1000 voip

destination-pattern 0.........

translate-outgoing called 1

session protocol sipv2

session target ipv4:<SERVER OF SIP PROVIDER>

!

sip-ua

retry invite 2

retry register 10

retry options 1

timers connect 100

sip-server dns:<SERVER OF SIP PROVIDER>

!

gatekeeper

shutdown

!

telephony-service

max-ephones 58

max-dn 192

ip source-address 172.16.10.2 port 2000

cnf-file location flash:

cnf-file perphone

user-locale U2 load CME-locale-cz_CZ-Czech-7.0.1.1.tar

user-locale 1 U1 cs

network-locale 1 U1

max-conferences 8 gain -6

moh music-on-hold.au

transfer-system full-consult

transfer-pattern 2..

create cnf-files version-stamp Jan 01 2002 00:00:00

!

ephone-template 1

!

ephone-dn 1

number 200 no-reg both

!

ephone-dn 2

number 200 no-reg both

!

ephone-dn 3

number 201 no-reg both

!

ephone-dn 4

number 201 no-reg both

!

ephone-dn 5

number 202 secondary XXXXXXXXX no-reg primary

!

ephone-dn 6

number 202 no-reg both

!

ephone-dn 7

number 203 no-reg both

!

ephone-dn 8

number 203 no-reg both

!

ephone-dn 9

number 204 no-reg both

!

ephone-dn 10

number 204 no-reg both

!

ephone 1

device-security-mode none

mac-address 1833.9D15.A7A6

ephone-template 1

type 7945

button 1:1 2:2

!

ephone 2

device-security-mode none

mac-address 10BD.1801.71ED

ephone-template 1

type 7945

button 1:3 2:4

!

ephone 3

device-security-mode none

mac-address 1833.9D14.0874

ephone-template 1

type 7945

button 1:5 2:6

!

ephone 4

device-security-mode none

mac-address 10BD.1800.084E

ephone-template 1

type 7945

button 1:7 2:8

!

ephone 5

device-security-mode none

mac-address 10BD.1800.4800

ephone-template 1

codec g729r8

type 7945

button 1:9 2:10

!

line con 0

logging synchronous

login local

line aux 0

line vty 0 4

session-timeout 2

exec-timeout 0 0

login local

transport input ssh

line vty 5 15

session-timeout 2

exec-timeout 0 0

login local

transport input ssh

!

scheduler allocate 20000 1000

end
====================================================

Output from debug ccsip error:

Nov 24 22:12:13.527: //667/D5699FDA87A1/SIP/Error/sipSPI_ipip_update_codec_params_in_channelInfo:

failed to update call entry

Nov 24 22:12:13.527: //667/D5699FDA87A1/SIP/Error/sipSPI_ipip_update_codec_params_in_channelInfo:

failed to update call entry

Nov 24 22:12:13.527: //667/D5699FDA87A1/SIP/Error/sipSPI_ipip_update_codec_params_in_channelInfo:

failed to update call entry

Nov 24 22:12:13.527: //667/D5699FDA87A1/SIP/Error/sipSPI_ipip_upd_2833_dtmf_params: Unable to find proper instance for FMTP

SIP: (667) fmtp attribute, level 1 instance 0 not found.

Nov 24 22:12:13.527: //667/D5699FDA87A1/SIP/Error/sipSPI_ipip_upd_2833_dtmf_params: Unable to acquire event mask for rfc2833

Nov 24 22:12:13.527: //667/D5699FDA87A1/SIP/Error/sipSPI_ipip_update_call_entry:

failed to update call entry

Nov 24 22:12:13.527: //-1/xxxxxxxxxxxx/SIP/Error/sipSPI_ipip_set_channel_count: Unable to set CHANNEL_COUNT for callid 667

Nov 24 22:12:13.527: //667/D5699FDA87A1/SIP/Error/sip_iwf_sip_copy_sdp_to_channelInfo: Channel count is not set at this point. Not SIP-SIP or SET_MODE is not done.

SIP: (668) Attribute mid, level 1 instance 1 not found.

Nov 24 22:12:14.027: //668/D5B5ECB587A6/SIP/Error/sipSPI_ipip_update_codec_params_in_channelInfo:

failed to update call entry

Nov 24 22:12:14.027: //668/D5B5ECB587A6/SIP/Error/sipSPI_ipip_update_codec_params_in_channelInfo:

failed to update call entry

Nov 24 22:12:14.027: //668/D5B5ECB587A6/SIP/Error/sipSPI_ipip_update_codec_params_in_channelInfo:

failed to update call entry

Nov 24 22:12:14.027: //668/D5B5ECB587A6/SIP/Error/sipSPI_ipip_upd_2833_dtmf_params: Unable to find proper instance for FMTP

SIP: (668) fmtp attribute, level 1 instance 0 not found.

Nov 24 22:12:14.027: //668/D5B5ECB587A6/SIP/Error/sipSPI_ipip_upd_2833_dtmf_params: Unable to acquire event mask for rfc2833

Nov 24 22:12:14.027: //668/D5B5ECB587A6/SIP/Error/sipSPI_ipip_update_call_entry:

failed to update call entry

Nov 24 22:12:14.027: //-1/xxxxxxxxxxxx/SIP/Error/sipSPI_ipip_set_channel_count: Unable to set CHANNEL_COUNT for callid 668

Nov 24 22:12:14.027: //668/D5B5ECB587A6/SIP/Error/sip_iwf_sip_copy_sdp_to_channelInfo: Channel count is not set at this point. Not SIP-SIP or SET_MODE is not done.

SIP: (669) Attribute mid, level 1 instance 1 not found.

Nov 24 22:12:15.111: //669/D65BF19F87AB/SIP/Error/sipSPI_ipip_update_codec_params_in_channelInfo:

failed to update call entry

Nov 24 22:12:15.111: //669/D65BF19F87AB/SIP/Error/sipSPI_ipip_update_codec_params_in_channelInfo:

failed to update call entry

Nov 24 22:12:15.111: //669/D65BF19F87AB/SIP/Error/sipSPI_ipip_update_codec_params_in_channelInfo:

failed to update call entry

Nov 24 22:12:15.111: //669/D65BF19F87AB/SIP/Error/sipSPI_ipip_upd_2833_dtmf_params: Unable to find proper instance for FMTP

SIP: (669) fmtp attribute, level 1 instance 0 not found.

Nov 24 22:12:15.111: //669/D65BF19F87AB/SIP/Error/sipSPI_ipip_upd_2833_dtmf_params: Unable to acquire event mask for rfc2833

Nov 24 22:12:15.111: //669/D65BF19F87AB/SIP/Error/sipSPI_ipip_update_call_entry:

failed to update call entry

Nov 24 22:12:15.111: //-1/xxxxxxxxxxxx/SIP/Error/sipSPI_ipip_set_channel_count: Unable to set CHANNEL_COUNT for callid 669

Nov 24 22:12:15.115: //669/D65BF19F87AB/SIP/Error/sip_iwf_sip_copy_sdp_to_channelInfo: Channel count is not set at this point. Not SIP-SIP or SET_MODE is not done.

SIP: (670) Attribute mid, level 1 instance 1 not found.

Nov 24 22:12:17.027: //670/D77FB29787B0/SIP/Error/sipSPI_ipip_update_codec_params_in_channelInfo:

failed to update call entry

Nov 24 22:12:17.027: //670/D77FB29787B0/SIP/Error/sipSPI_ipip_update_codec_params_in_channelInfo:

failed to update call entry

Nov 24 22:12:17.027: //670/D77FB29787B0/SIP/Error/sipSPI_ipip_update_codec_params_in_channelInfo:

failed to update call entry

Nov 24 22:12:17.027: //670/D77FB29787B0/SIP/Error/sipSPI_ipip_upd_2833_dtmf_params: Unable to find proper instance for FMTP

SIP: (670) fmtp attribute, level 1 instance 0 not found.

Nov 24 22:12:17.027: //670/D77FB29787B0/SIP/Error/sipSPI_ipip_upd_2833_dtmf_params: Unable to acquire event mask for rfc2833

Nov 24 22:12:17.027: //670/D77FB29787B0/SIP/Error/sipSPI_ipip_update_call_entry:

failed to update call entry

Nov 24 22:12:17.027: //-1/xxxxxxxxxxxx/SIP/Error/sipSPI_ipip_set_channel_count: Unable to set CHANNEL_COUNT for callid 670

Nov 24 22:12:17.027: //670/D77FB29787B0/SIP/Error/sip_iwf_sip_copy_sdp_to_channelInfo: Channel count is not set at this point. Not SIP-SIP or SET_MODE is not done.

voip-router#

SIP: (671) Attribute mid, level 1 instance 1 not found.

Nov 24 22:12:21.027: //671/D9E2ABBA87B5/SIP/Error/sipSPI_ipip_update_codec_params_in_channelInfo:

failed to update call entry

Nov 24 22:12:21.027: //671/D9E2ABBA87B5/SIP/Error/sipSPI_ipip_update_codec_params_in_channelInfo:

failed to update call entry

Nov 24 22:12:21.027: //671/D9E2ABBA87B5/SIP/Error/sipSPI_ipip_update_codec_params_in_channelInfo:

failed to update call entry

Nov 24 22:12:21.027: //671/D9E2ABBA87B5/SIP/Error/sipSPI_ipip_upd_2833_dtmf_params: Unable to find proper instance for FMTP

SIP: (671) fmtp attribute, level 1 instance 0 not found.

Nov 24 22:12:21.027: //671/D9E2ABBA87B5/SIP/Error/sipSPI_ipip_upd_2833_dtmf_params: Unable to acquire event mask for rfc2833

Nov 24 22:12:21.027: //671/D9E2ABBA87B5/SIP/Error/sipSPI_ipip_update_call_entry:

failed to update call entry

Nov 24 22:12:21.027: //-1/xxxxxxxxxxxx/SIP/Error/sipSPI_ipip_set_channel_count: Unable to set CHANNEL_COUNT for callid 671

Nov 24 22:12:21.027: //671/D9E2ABBA87B5/SIP/Error/sip_iwf_sip_copy_sdp_to_channelInfo: Channel count is not set at this point. Not SIP-SIP or SET_MODE is not done.

SIP: (672) Attribute mid, level 1 instance 1 not found.

Nov 24 22:12:25.031: //672/DC45A4DD87BA/SIP/Error/sipSPI_ipip_update_codec_params_in_channelInfo:

failed to update call entry

Nov 24 22:12:25.031: //672/DC45A4DD87BA/SIP/Error/sipSPI_ipip_update_codec_params_in_channelInfo:

failed to update call entry

Nov 24 22:12:25.031: //672/DC45A4DD87BA/SIP/Error/sipSPI_ipip_update_codec_params_in_channelInfo:

failed to update call entry

Nov 24 22:12:25.031: //672/DC45A4DD87BA/SIP/Error/sipSPI_ipip_upd_2833_dtmf_params: Unable to find proper instance for FMTP

SIP: (672) fmtp attribute, level 1 instance 0 not found.

Nov 24 22:12:25.031: //672/DC45A4DD87BA/SIP/Error/sipSPI_ipip_upd_2833_dtmf_params: Unable to acquire event mask for rfc2833

Nov 24 22:12:25.031: //672/DC45A4DD87BA/SIP/Error/sipSPI_ipip_update_call_entry:

failed to update call entry

Nov 24 22:12:25.031: //-1/xxxxxxxxxxxx/SIP/Error/sipSPI_ipip_set_channel_count: Unable to set CHANNEL_COUNT for callid 672

Nov 24 22:12:25.031: //672/DC45A4DD87BA/SIP/Error/sip_iwf_sip_copy_sdp_to_channelInfo: Channel count is not set at this point. Not SIP-SIP or SET_MODE is not done.

voip-router#

SIP: (673) Attribute mid, level 1 instance 1 not found.

Nov 24 22:12:29.035: //673/DEA801D887BF/SIP/Error/sipSPI_ipip_update_codec_params_in_channelInfo:

failed to update call entry

Nov 24 22:12:29.035: //673/DEA801D887BF/SIP/Error/sipSPI_ipip_update_codec_params_in_channelInfo:

failed to update call entry

Nov 24 22:12:29.035: //673/DEA801D887BF/SIP/Error/sipSPI_ipip_update_codec_params_in_channelInfo:

failed to update call entry

Nov 24 22:12:29.035: //673/DEA801D887BF/SIP/Error/sipSPI_ipip_upd_2833_dtmf_params: Unable to find proper instance for FMTP

SIP: (673) fmtp attribute, level 1 instance 0 not found.

Nov 24 22:12:29.035: //673/DEA801D887BF/SIP/Error/sipSPI_ipip_upd_2833_dtmf_params: Unable to acquire event mask for rfc2833

Nov 24 22:12:29.035: //673/DEA801D887BF/SIP/Error/sipSPI_ipip_update_call_entry:

failed to update call entry

Nov 24 22:12:29.035: //-1/xxxxxxxxxxxx/SIP/Error/sipSPI_ipip_set_channel_count: Unable to set CHANNEL_COUNT for callid 673

Nov 24 22:12:29.035: //673/DEA801D887BF/SIP/Error/sip_iwf_sip_copy_sdp_to_channelInfo: Channel count is not set at this point. Not SIP-SIP or SET_MODE is not done.

Ayodeji Okanlawon · ‎11-24-2012

Standa,

Can you try and add the ff:

dial-peer voice 1 voip

incoming called number .

session protocol sipv2

dtmf-relay rtp-nte

Please do a test call after adding this command and send the ff:

debug ccsip messages..

Please send the calling and called number

Please rate all useful posts

"'Nature is too thin a screen, the glory of the omnipresent God bursts through it everywhere"-Ralph Waldo Emerson

Please rate all useful posts

zittastanislav · ‎11-24-2012

Thank you for your answer, sir. I've added config you mentioned, but nothing changed. Still cannot hear ringback on my cell phone and don't know if sccp phone (under ephone 3)

is ringing. Here is output from debug ccsip messages: I had to change real dialed number to XXXXXXXXX, but it is same number (secondary) as in ephone-dn 5....

===================Output from debug ccsip messages:

Received:

INVITE sip:XXXXXXXXX@

;user=phone SIP/2.0

Via: SIP/2.0/UDP 88.103.241.253:5060;branch=z9hG4bKbeivk9lmhkkvn03d55iaepv9p4

To: <>XXXXXXXXX@

;user=phone>

From: <777912354>;user=phone>;tag=3cae2031c4b9d5867b7d086a0417ed4c

Call-ID: ERICSSONBTK_TERM_5f4826e753a3651cb704842b905c042e@10.254.1.53

CSeq: 61254281 INVITE

Max-Forwards: 68

Content-Length: 235

Contact: <88.103.241.253:5060>

Content-Type: application/sdp

Allow: INVITE, CANCEL, ACK, BYE

Accept: application/sdp

P-Asserted-Identity: <777912354>:5060;user=phone>

Privacy: none

P-Charging-Vector: icid-value="50b14c6879671201351887";icid-generated-at=10.254.0.132;ericsson-imt=1;oaid="BOT1B7";orig-ioi=

Session-Expires: 1800

Min-SE: 1800

v=0

o=- 1353796712 1353796712 IN IP4 88.103.241.253

s=Basic Session

c=IN IP4 88.103.241.253

t=0 0

m=audio 50176 RTP/AVP 18 0 8 99 13

a=fmtp:18 annexb=no;mr=0;mg=0;jmdelay=no;mrmods=

a=rtpmap:99 telephone-event/8000

a=ptime:20

SIP: (678) Attribute mid, level 1 instance 1 not found.

Nov 24 22:38:32.329: //678/82745F7D87D8/SIP/Error/sipSPI_ipip_update_codec_params_in_channelInfo:

failed to update call entry

Nov 24 22:38:32.329: //678/82745F7D87D8/SIP/Error/sipSPI_ipip_upd_2833_dtmf_params: Unable to find proper instance for FMTP

SIP: (678) fmtp attribute, level 1 instance 0 not found.

Nov 24 22:38:32.329: //678/82745F7D87D8/SIP/Error/sipSPI_ipip_upd_2833_dtmf_params: Unable to acquire event mask for rfc2833

Nov 24 22:38:32.329: //678/82745F7D87D8/SIP/Error/sipSPI_ipip_update_call_entry:

failed to update call entry

Nov 24 22:38:32.329: //-1/xxxxxxxxxxxx/SIP/Error/sipSPI_ipip_set_channel_count: Unable to set CHANNEL_COUNT for callid 678

Nov 24 22:38:32.329: //678/82745F7D87D8/SIP/Error/sip_iwf_sip_copy_sdp_to_channelInfo: Channel count is not set at this point. Not SIP-SIP or SET_MODE is not done.

Nov 24 22:38:32.345: //678/82745F7D87D8/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 88.103.241.253:5060;branch=z9hG4bKbeivk9lmhkkvn03d55iaepv9p4

From: <777912354>;user=phone>;tag=3cae2031c4b9d5867b7d086a0417ed4c

To: ;user=phone>

Date: Sat, 24 Nov 2012 22:38:32 GMT

Call-ID: ERICSSONBTK_TERM_5f4826e753a3651cb704842b905c042e@10.254.1.53

CSeq: 61254281 INVITE

Allow-Events: telephone-event

Server: Cisco-SIPGateway/IOS-12.x

Content-Length: 0

Nov 24 22:38:32.349: //-1/xxxxxxxxxxxx/SIP/Error/ccsip_call_service_msg: ccb NULL, unable to update the callinfo ui parameters

Nov 24 22:38:32.353: //678/82745F7D87D8/SIP/Error/sipSPI_ipip_set_history_info_header: Not SIP2SIP mode

Nov 24 22:38:32.353: //678/82745F7D87D8/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 180 Ringing

Via: SIP/2.0/UDP 88.103.241.253:5060;branch=z9hG4bKbeivk9lmhkkvn03d55iaepv9p4

From: <777912354>;user=phone>;tag=3cae2031c4b9d5867b7d086a0417ed4c

To: ;user=phone>;tag=70552BC-1ADD

Date: Sat, 24 Nov 2012 22:38:32 GMT

Call-ID: ERICSSONBTK_TERM_5f4826e753a3651cb704842b905c042e@10.254.1.53

CSeq: 61254281 INVITE

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

Allow-Events: telephone-event

Remote-Party-ID: <202>;party=called;screen=no;privacy=off

Contact:

Server: Cisco-SIPGateway/IOS-12.x

Content-Length: 0

Nov 24 22:38:32.825: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

INVITE sip:XXXXXXXXX@;user=phone SIP/2.0

Via: SIP/2.0/UDP 88.103.241.253:5060;branch=z9hG4bKbeivk9lmhkkvn03d55iaepv9p4

To: ;user=phone>

From: <777912354>;user=phone>;tag=3cae2031c4b9d5867b7d086a0417ed4c

Call-ID: ERICSSONBTK_TERM_5f4826e753a3651cb704842b905c042e@10.254.1.53

CSeq: 61254281 INVITE

Max-Forwards: 68

Content-Length: 235

Contact: <88.103.241.253:5060>

Content-Type: application/sdp

Allow: INVITE, CANCEL, ACK, BYE

Accept: application/sdp

P-Asserted-Identity: <777912354>:5060;user=phone>

Privacy: none

P-Charging-Vector: icid-value="50b14c6879671201351887";icid-generated-at=10.254.0.132;ericsson-imt=1;oaid="BOT1B7";orig-ioi=

Session-Expires: 1800

Min-SE: 1800

v=0

o=- 1353796712 1353796712 IN IP4 88.103.241.253

s=Basic Session

c=IN IP4 88.103.241.253

t=0 0

m=audio 50176 RTP/AVP 18 0 8 99 13

a=fmtp:18 annexb=no;mr=0;mg=0;jmdelay=no;mrmods=

a=rtpmap:99 telephone-event/8000

a=ptime:20

Nov 24 22:38:32.829: //678/82745F7D87D8/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 180 Ringing

Via: SIP/2.0/UDP 88.103.241.253:5060;branch=z9hG4bKbeivk9lmhkkvn03d55iaepv9p4

From: <777912354>;user=phone>;tag=3cae2031c4b9d5867b7d086a0417ed4c

To: ;user=phone>;tag=70552BC-1ADD

Date: Sat, 24 Nov 2012 22:38:32 GMT

Call-ID: ERICSSONBTK_TERM_5f4826e753a3651cb704842b905c042e@10.254.1.53

CSeq: 61254281 INVITE

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

Allow-Events: telephone-event

Remote-Party-ID: <202>;party=called;screen=no;privacy=off

Contact:

Server: Cisco-SIPGateway/IOS-12.x

Content-Length: 0

Nov 24 22:38:33.825: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

INVITE sip:XXXXXXXXX@;user=phone SIP/2.0

Via: SIP/2.0/UDP 88.103.241.253:5060;branch=z9hG4bKbeivk9lmhkkvn03d55iaepv9p4

To: ;user=phone>

From: <777912354>;user=phone>;tag=3cae2031c4b9d5867b7d086a0417ed4c

Call-ID: ERICSSONBTK_TERM_5f4826e753a3651cb704842b905c042e@10.254.1.53

CSeq: 61254281 INVITE

Max-Forwards: 68

Content-Length: 235

Contact: <88.103.241.253:5060>

Content-Type: application/sdp

Allow: INVITE, CANCEL, ACK, BYE

Accept: application/sdp

P-Asserted-Identity: <777912354>:5060;user=phone>

Privacy: none

P-Charging-Vector: icid-value="50b14c6879671201351887";icid-generated-at=10.254.0.132;ericsson-imt=1;oaid="BOT1B7";orig-ioi=

Session-Expires: 1800

Min-SE: 1800

v=0

o=- 1353796712 1353796712 IN IP4 88.103.241.253

s=Basic Session

c=IN IP4 88.103.241.253

t=0 0

m=audio 50176 RTP/AVP 18 0 8 99 13

a=fmtp:18 annexb=no;mr=0;mg=0;jmdelay=no;mrmods=

a=rtpmap:99 telephone-event/8000

a=ptime:20

Thank you for your time,

Standa

zittastanislav · ‎11-25-2012

Maybe I've discovered another problem. Now, i have only one dial peer (just for simplicity) and trying to make outbound call. Dial-peer now looks like this:

dial-peer voice 1 voip

description Outbound

destination-pattern .T

session protocol sipv2

session target ipv4:10.5.5.1

session transport udp

no vad
When i try call to outside number, sip messages looks like this:

Nov 25 14:29:42.717: //812/630C9707898E/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 192.168.120.240:5060;branch=z9hG4bK-d8754z-740b7b9297720ac3-1---d8754z-;rport

From: <100>;tag=6f5dfdd4

To: <736185542>

Date: Sun, 25 Nov 2012 14:29:42 GMT

Call-ID: NDhiNzIwOWY0MDNmZGI4OGExZGIxMzc0Mjg2YzM2NzU.

CSeq: 1 INVITE

Allow-Events: telephone-event

Server: Cisco-SIPGateway/IOS-12.x

Content-Length: 0

Nov 25 14:29:42.721: //812/630C9707898E/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 503 Service Unavailable

Via: SIP/2.0/UDP 192.168.120.240:5060;branch=z9hG4bK-d8754z-740b7b9297720ac3-1---d8754z-;rport

From: <100>;tag=6f5dfdd4

To: <736185542>;tag=A6C218C-51F

Date: Sun, 25 Nov 2012 14:29:42 GMT

Call-ID: NDhiNzIwOWY0MDNmZGI4OGExZGIxMzc0Mjg2YzM2NzU.

CSeq: 1 INVITE

Allow-Events: telephone-event

Server: Cisco-SIPGateway/IOS-12.x

Reason: Q.850;cause=47

Content-Length: 0

Nov 25 14:29:42.725: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

ACK sip:736185542@172.16.10.2 SIP/2.0

Via: SIP/2.0/UDP 192.168.120.240:5060;branch=z9hG4bK-d8754z-740b7b9297720ac3-1---d8754z-;rport

Max-Forwards: 70

To: <736185542>72.16.10.2>;tag=A6C218C-51F

From: <100>;tag=6f5dfdd4

Call-ID: NDhiNzIwOWY0MDNmZGI4OGExZGIxMzc0Mjg2YzM2NzU.

CSeq: 1 ACK

Content-Length: 0

Personally. i think that call is incorrectly routed. 172.16.10.2 is interface of CME. Call isn't going to session target configured in dial-peer. Instead of this, call goes to CME. This behavior isn't understandable for me. Is dial-peer for outbound calls (1) incorrectly configured? Is there anything crucial missing in my config?

Best regards,

Standa

Chris Deren · ‎11-25-2012

Please bind your SIP to proper interfaces, you can either do this under the dial-peers such as this:

voice service voip

sip

bind control source-interface GigabitEthernet0/0

bind media source-interface GigabitEthernet0/0

or dial-peer:

dial-peer voice 1 voip

voice-class sip bind control source-interface GigabitEthernet0/1

voice-class sip bind media source-interface GigabitEthernet0/1

Though "SIP/2.0 503 Service Unavailable" indicates the SIP service is not available, so ensure you are pointing to proper destination and port, and that no NAT is needed.

HTH,

Chris

zittastanislav · ‎11-25-2012

Thank you Chris,

I tried steps you described above, but still no luck - still cannot hear ringback when trying inbound call and still getting 503 service unavailable when trying outbound call.

Let me ask you few questions:

In dial-peer voice 1 ... Should i choose interface that faces VoIP provider?
In global configoration of voip - should i choose interface that faces my internal network?

If i should describe my topology in short: There is CME 8.6 sitting in between my internal network and voice provider and internet.

Ge 0/1 has IP address 172.16.10.2, Ge 0/0 has 10.5.5.2.

Gateway to VoIP provider has IP address 10.5.5.1 and gateway to internet has 172.16.10.1. In routing table, i have static route saying: Everythig going to SIP server (ip address 88.103.xxx.xx) shoud go over 10.5.5.1.

Thank you for your precious time,

Standa

Ayodeji Okanlawon · ‎11-25-2012

Looking at the logs...

The reason for service unavailable is given as cause code 47... This is usually codec related or media capability mismatch..

SIP/2.0 503 Service Unavailable

Via: SIP/2.0/UDP 192.168.120.240:5060;branch=z9hG4bK-d8754z-740b7b9297720ac3-1---d8754z-;rport

From: <100>;tag=6f5dfdd4

To: <736185542>;tag=A6C218C-51F

Date: Sun, 25 Nov 2012 14:29:42 GMT

Call-ID: NDhiNzIwOWY0MDNmZGI4OGExZGIxMzc0Mjg2YzM2NzU.

CSeq: 1 INVITE

Allow-Events: telephone-event

Server: Cisco-SIPGateway/IOS-12.x

Reason: Q.850;cause=47

Content-Length: 0

Can you try this...

conf t

voice class codec 1

codec preference 1 g729r8

codec preference 2 g711ulaw

codec preference 3 g711alaw

Also change your dial-peer as follows

dial-peer voice 1 voip

voice-class codec 1

description Outbound

destination-pattern .T

session transport tcp

session protocol sipv2

session target ipv4:88.103.241.253

no vad

What is the ip address of your sip provider? Is it not 88.103.241.253? why are you sending your traffic to 10.5.5.1?

I think you should send it to the ip address of the sip provider as I have put in the config..

Do a test call again and send the

debug ccsip messages

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Please rate all useful posts

zittastanislav · ‎11-26-2012

Good afternoon,

first of all - sorry for replying after a day. I was quite busy.

Finally, I figured out where was problem(s):

sip control and media interface has to be the one connecting me to ITSP provider - in my case, it's Gi 0/0
There were completely missing translation profiles. In other words - Outgoing call sent local extension to ITSP. Naturally, ITSP rejected call originating from local extension (for example extension 296).

I'm complete newbie in VoIP network, but my goal is to learn more. I've learn't from "CCNA Voice official certification guide", where SIP and VoIP isn't described rigorously. Therefore, I'd like to ask you which good book / web tutorial / video tutorial focused on VoIP technology and SIP and Cisco would you recommend.

Thank you all for your effort to help and for your time,

Standa

Hafthor Hilmarsson O'Connor · ‎12-17-2012

Yes you will have to source the traffic towards your SIP provider with the public ip address ( or and ip address that towards him )

then source the dial-peer that is towards your CUCM with the network that is on that interface.

If you are using for example backup interfaces over DMVPN or 3G you would need to make a loopback and then source the traffic from there.

The commands that are listed above are correct for 15.X Ios

That is

voice-class source interface loopback 1

if its IOS 12.4 then its

voice-class sip bind control source-interface

voice-class sip bind media source-interface

Regards what the best book , its been some time since I read any ccna related material, however , the official guide that Jeremy Cioara and Michael Valentine wrote is a good one.

Here is a link

http://www.ciscopress.com/store/ccna-voice-640-461-official-cert-guide-9781587204173

I then recomend you go for CCVP

CCIE-Collaboration #24527