11-24-2012 02:13 PM - edited 03-16-2019 02:22 PM
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 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
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 ....
<USERNAMES AND PASSWORDS>
!
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
!
!
<ROUTING TABLE ITEMS - OUTPUT OMMITED>
!
!
!
!
!
!
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.
11-24-2012 02:26 PM
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
11-24-2012 02:46 PM
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=3cae2031c4b9d5867b7d086a0417ed4c777912354>
Call-ID: ERICSSONBTK_TERM_5f4826e753a3651cb704842b905c042e@10.254.1.53
CSeq: 61254281 INVITE
Max-Forwards: 68
Content-Length: 235
Contact: <88.103.241.253:5060>88.103.241.253:5060>
Content-Type: application/sdp
Allow: INVITE, CANCEL, ACK, BYE
Accept: application/sdp
P-Asserted-Identity: <777912354>:5060;user=phone>777912354>
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=3cae2031c4b9d5867b7d086a0417ed4c777912354>
To:
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=3cae2031c4b9d5867b7d086a0417ed4c777912354>
To:
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=off202>
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@
Via: SIP/2.0/UDP 88.103.241.253:5060;branch=z9hG4bKbeivk9lmhkkvn03d55iaepv9p4
To:
From: <777912354>;user=phone>;tag=3cae2031c4b9d5867b7d086a0417ed4c777912354>
Call-ID: ERICSSONBTK_TERM_5f4826e753a3651cb704842b905c042e@10.254.1.53
CSeq: 61254281 INVITE
Max-Forwards: 68
Content-Length: 235
Contact: <88.103.241.253:5060>88.103.241.253:5060>
Content-Type: application/sdp
Allow: INVITE, CANCEL, ACK, BYE
Accept: application/sdp
P-Asserted-Identity: <777912354>:5060;user=phone>777912354>
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=3cae2031c4b9d5867b7d086a0417ed4c777912354>
To:
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=off202>
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@
Via: SIP/2.0/UDP 88.103.241.253:5060;branch=z9hG4bKbeivk9lmhkkvn03d55iaepv9p4
To:
From: <777912354>;user=phone>;tag=3cae2031c4b9d5867b7d086a0417ed4c777912354>
Call-ID: ERICSSONBTK_TERM_5f4826e753a3651cb704842b905c042e@10.254.1.53
CSeq: 61254281 INVITE
Max-Forwards: 68
Content-Length: 235
Contact: <88.103.241.253:5060>88.103.241.253:5060>
Content-Type: application/sdp
Allow: INVITE, CANCEL, ACK, BYE
Accept: application/sdp
P-Asserted-Identity: <777912354>:5060;user=phone>777912354>
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
11-25-2012 06:33 AM
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=6f5dfdd4100>
To: <736185542>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=6f5dfdd4100>
To: <736185542>;tag=A6C218C-51F736185542>
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-51F736185542>
From: <100>;tag=6f5dfdd4100>
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
11-25-2012 07:52 AM
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
11-25-2012 09:30 AM
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:
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
11-25-2012 10:54 AM
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=6f5dfdd4100>
To: <736185542>;tag=A6C218C-51F736185542>
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
Please rate all useful posts
"'Nature is too thin a screen, the glory of the omnipresent God bursts through it everywhere"-Ralph Waldo Emerson
11-26-2012 09:05 AM
Good afternoon,
first of all - sorry for replying after a day. I was quite busy.
Finally, I figured out where was problem(s):
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
12-17-2012 07:08 AM
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
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