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CUBE- Incoming call and translation rule

Thiago Cella
Beginner
Beginner

Hi,

 

Im trying implement translation rule, when someone from outside call 551130741111, the cube have to redirect for internal extension : 399.

 

I configured the follow parameters, but it is not working, could you help me? Tks

 

voice translation-rule 2
rule 1 /551130741111/ /399/
!
!
voice translation-profile SIP-IN
translate called 2
!

dial-peer voice 1002 voip
translation-profile incoming SIP-IN
session protocol sipv2
session target ipv4:10.255.240.111
incoming called-number .%
voice-class sip dtmf-relay force rtp-nte
voice-class sip bind control source-interface GigabitEthernet0/2
voice-class sip bind media source-interface GigabitEthernet0/2
dtmf-relay sip-info sip-kpml sip-notify rtp-nte
codec g711ulaw
!

Follow the debug ccsip messages, when i make a test:

 

 

005765: *Aug 11 01:17:09.999: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:551130741111@192.168.0.100 SIP/2.0
Max-Forwards: 64
Supported: 100rel
To: <sip:+55551130741111@10.255.241.68>
From: <sip:+5511976637000@10.255.241.68>;tag=3742940362-1679752367
P-Asserted-Identity: <sip:976637000@ims4.vivo.net.br>
Call-ID: 11727671-3742940362-604125931@sip.domain.com
CSeq: 1 INVITE
Allow: PUBLISH,MESSAGE,UPDATE,PRACK,SUBSCRIBE,REFER,INFO,NOTIFY,REGISTER,OPTIONS,BYE,INVITE,ACK,CANCEL
Via: SIP/2.0/UDP 10.255.241.68:5060;branch=z9hG4bK85f8394474a6584a152de5aff0a3ae2c
Contact: <sip:+5511976637000@10.255.241.68:5060>
Content-Type: application/sdp
Accept: application/sdp
P-Charging-Vector: icid-value=mgcf--20180810223921-100320901;orig-ioi=243;term-ioi=3239
P-Notification: caller-control
Content-Length: 216

v=0
o=ngn-br-spo-be-sbc1 202625063 202625063 IN IP4 10.255.241.68
s=sip call
c=IN IP4 10.255.241.76
t=0 0
m=audio 37090 RTP/AVP 8 100
a=rtpmap:100 telephone-event/8000
a=fmtp:100 0-15
a=ptime:20
a=sendrecv

005766: *Aug 11 01:17:10.007: //21954/1AB6D7868E0F/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.255.241.68:5060;branch=z9hG4bK85f8394474a6584a152de5aff0a3ae2c
From: <sip:+5511976637000@10.255.241.68>;tag=3742940362-1679752367
To: <sip:+55551130741111@10.255.241.68>
Date: Sat, 11 Aug 2018 01:17:09 GMT
Call-ID: 11727671-3742940362-604125931@sip.domain.com
CSeq: 1 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-15.5.3.M
Content-Length: 0


005767: *Aug 11 01:17:10.007: //21954/1AB6D7868E0F/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 500 Internal Server Error
Via: SIP/2.0/UDP 10.255.241.68:5060;branch=z9hG4bK85f8394474a6584a152de5aff0a3ae2c
From: <sip:+5511976637000@10.255.241.68>;tag=3742940362-1679752367
To: <sip:+55551130741111@10.255.241.68>;tag=EF2D69DC-182D
Call-ID: 11727671-3742940362-604125931@sip.domain.com
CSeq: 1 INVITE
Content-Length: 0


005768: *Aug 11 01:17:10.011: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:551130741111@192.168.0.100 SIP/2.0
Max-Forwards: 70
To: <sip:+55551130741111@10.255.241.68>;tag=EF2D69DC-182D
From: <sip:+5511976637000@10.255.241.68>;tag=3742940362-1679752367
Call-ID: 11727671-3742940362-604125931@sip.domain.com
CSeq: 1 ACK
Via: SIP/2.0/UDP 10.255.241.68:5060;branch=z9hG4bK85f8394474a6584a152de5aff0a3ae2c
Contact: <sip:+5511976637000@10.255.241.68:5060>
Content-Length: 0

10 Replies 10

Hi Thiago

On your voice translation rule add what follows 

 

voice translation-rule 2

rule 2 /.+55551130741111/ /399/

 

Please let me know 

 

HTH

 

Regards

Carlo

Please rate all helpful posts "The more you help the more you learn"

The call may be hitting another inbound dial-peer, causing the translation not to occur.

 

I recommend, during off-peak times, running "debug voip dialpeer inout" (do NOT use terminal monitor - use buffered logging instead - "logging buffered 1000000 debug") and then making a test call. The output will show you which inbound dial-peer is being hit. 

 

Ryan

Oleksandr Yurchenko
Rising star
Rising star

Hi Thiago Cella

Check next:

router# test voice translation-rule 2 551130741111

 

After also you can check which inbound dial-peer working. See output from:

debug voice ccapi inout

Your issue is not the translation rule. Your router is not even processing thr call request. You need to check why your router is not accepting the call. Does the ip address in tbe host portion of the RURI match any interface on your router?

Please rate all useful posts

+5 Deji,

I didn’t noticed the internal server error message :)

 

Cheers

 

Carlo

Please rate all helpful posts "The more you help the more you learn"

Also, a "debug ccsip messages" and "debug ccsip error" can help illuminate the cause of the 500 error.  

 

Ryan

Tks Friends, still dont working the translation.

I attached the config of gw, and the topology.Follow the details:

 

- As you can see the topology bellow, the communication beetween branch 1 and SIP provider, works by network 10.7.75.152/29, so i configured on gw of voice the IP: 10.7.75.154. The ISP only accept this address, so i have to configure a NAT to translate my gw voice from 192.168.0.100 to 10.7.7.154;

 

- the outboung calls are working;

 

 

tac.jpg

 

 

 

follow debug ccsip messages :

 

013322: *Aug 12 18:40:20.189: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:1130741111@192.168.0.100 SIP/2.0
Max-Forwards: 62
Supported: 100rel
To: <sip:+551130741111@10.255.241.68>
From: <sip:+5511976637000@10.255.241.68>;tag=3743089353-693987752
P-Asserted-Identity: <sip:976637000@sip2.domain2.com>
Call-ID: 12119249-3743089353-233704787@sip.domain.com
CSeq: 1 INVITE
Allow: PUBLISH,MESSAGE,UPDATE,PRACK,SUBSCRIBE,REFER,INFO,NOTIFY,REGISTER,OPTIONS,BYE,INVITE,ACK,CANCEL
Via: SIP/2.0/UDP 10.255.241.68:5060;branch=z9hG4bK1ffa0cb343a576111d4ffb0e3676332e
Contact: <sip:+5511976637000@10.255.241.68:5060>
Content-Type: application/sdp
Accept: application/sdp
P-Charging-Vector: icid-value=mgcf--20180812160233-100420855;orig-ioi=230;term-ioi=3239
P-Notification: caller-control
Content-Length: 241

v=0
o=ngn-br-spo-be-sbc1 201944986 201944986 IN IP4 10.255.241.68
s=sip call
c=IN IP4 10.255.241.76
t=0 0
m=audio 38300 RTP/AVP 18 8 100
a=fmtp:18 annexb=yes
a=rtpmap:100 telephone-event/8000
a=fmtp:100 0-15
a=ptime:20
a=sendrecv

013323: *Aug 12 18:40:20.197: //24867/FFC99F669C7F/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.255.241.68:5060;branch=z9hG4bK1ffa0cb343a576111d4ffb0e3676332e
From: <sip:+5511976637000@10.255.241.68>;tag=3743089353-693987752
To: <sip:+551130741111@10.255.241.68>
Date: Sun, 12 Aug 2018 18:40:20 GMT
Call-ID: 12119249-3743089353-233704787@sip.domain.com
CSeq: 1 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-15.5.3.M
Content-Length: 0


013324: *Aug 12 18:40:20.197: //24868/FFC99F669C7F/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:1130741111@192.168.0.103:5060 SIP/2.0
Via: SIP/2.0/UDP 10.7.75.154:5060;branch=z9hG4bK1DCF97
From: <sip:976637000@10.7.75.154>;tag=F80ED2D4-818
To: <sip:1130741111@192.168.0.103>
Date: Sun, 12 Aug 2018 18:40:20 GMT
Call-ID: FFCAD7B6-9D9511E8-9C85BC10-BA50BE76@10.7.75.154
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 4291403622-2643792360-2625616912-3125853814
User-Agent: Cisco-SIPGateway/IOS-15.5.3.M
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1534099220
Contact: <sip:976637000@10.7.75.154:5060>
Call-Info: <sip:10.7.75.154:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
Expires: 180
Allow-Events: kpml, telephone-event
Max-Forwards: 61
P-Asserted-Identity: <sip:976637000@10.7.75.154>
P-Charging-Vector: icid-value=mgcf--20180812160233-100420855;orig-ioi=230;term-ioi=3239
P-Notification: caller-control
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 239

v=0
o=ngn-br-spo-be-sbc1 201944986 201944986 IN IP4 10.255.241.68
s=sip call
c=IN IP4 10.7.75.154
t=0 0
m=audio 18632 RTP/AVP 18 8 100
a=fmtp:18 annexb=yes
a=rtpmap:100 telephone-event/8000
a=fmtp:100 0-15
a=ptime:20
a=sendrecv

013325: *Aug 12 18:40:20.201: //24868/FFC99F669C7F/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.7.75.154:5060;branch=z9hG4bK1DCF97
From: <sip:976637000@10.7.75.154>;tag=F80ED2D4-818
To: <sip:1130741111@192.168.0.103>
Date: Sun, 12 Aug 2018 19:02:33 GMT
Call-ID: FFCAD7B6-9D9511E8-9C85BC10-BA50BE76@10.7.75.154
CSeq: 101 INVITE
Allow-Events: presence
Content-Length: 0


013326: *Aug 12 18:40:20.201: //24868/FFC99F669C7F/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 10.7.75.154:5060;branch=z9hG4bK1DCF97
From: <sip:976637000@10.7.75.154>;tag=F80ED2D4-818
To: <sip:1130741111@192.168.0.103>;tag=1815572526
Date: Sun, 12 Aug 2018 19:02:33 GMT
Call-ID: FFCAD7B6-9D9511E8-9C85BC10-BA50BE76@10.7.75.154
CSeq: 101 INVITE
Allow-Events: presence
Warning: 399 CUCM "Unable to find a device handler for the request received on port 62001 from 10.7.75.154"
Content-Length: 0


013327: *Aug 12 18:40:20.205: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:1130741111@192.168.0.103:5060 SIP/2.0
Via: SIP/2.0/UDP 10.7.75.154:5060;branch=z9hG4bK1DCF97
From: <sip:976637000@10.7.75.154>;tag=F80ED2D4-818
To: <sip:1130741111@192.168.0.103>;tag=1815572526
Date: Sun, 12 Aug 2018 18:40:20 GMT
Call-ID: FFCAD7B6-9D9511E8-9C85BC10-BA50BE76@10.7.75.154
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: kpml, telephone-event
Content-Length: 0


013328: *Aug 12 18:40:20.205: //24867/FFC99F669C7F/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 500 Internal Server Error
Via: SIP/2.0/UDP 10.255.241.68:5060;branch=z9hG4bK1ffa0cb343a576111d4ffb0e3676332e
From: <sip:+5511976637000@10.255.241.68>;tag=3743089353-693987752
To: <sip:+551130741111@10.255.241.68>;tag=F80ED2DC-E65
Call-ID: 12119249-3743089353-233704787@sip.domain.com
CSeq: 1 INVITE
Content-Length: 0


013329: *Aug 12 18:40:20.213: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:1130741111@192.168.0.100 SIP/2.0
Max-Forwards: 70
To: <sip:+551130741111@10.255.241.68>;tag=F80ED2DC-E65
From: <sip:+5511976637000@10.255.241.68>;tag=3743089353-693987752
Call-ID: 12119249-3743089353-233704787@sip.domain.com
CSeq: 1 ACK
Via: SIP/2.0/UDP 10.255.241.68:5060;branch=z9hG4bK1ffa0cb343a576111d4ffb0e3676332e
Contact: <sip:+5511976637000@10.255.241.68:5060>
Content-Length: 0

 

 

 

follow debug ccsip error:

 

013330: *Aug 12 18:44:38.105: //-1/xxxxxxxxxxxx/SIP/Error/httpish_msg_free:
Freeing NULL pointer!
013331: *Aug 12 18:44:38.109: //-1/99847ABD9C8B/SIP/Error/sipSPI_ipip_build_consolidated_header_list:
No headers associated with passthrulist tag: 0 and copylist tag: 0
SIP: Group (a= group line) attribute, level 65535 instance 1 not found.
SIP: Attribute mid, level 1 instance 1 not found.
013332: *Aug 12 18:44:38.113: //24875/99847ABD9C8B/SIP/Error/sipSPIGetCallServerGroupTargets:
No server group configured
013333: *Aug 12 18:44:38.113: //24875/99847ABD9C8B/SIP/Error/sipSPI_ipip_build_consolidated_header_list:
No headers associated with passthrulist tag: 0 and copylist tag: 0
013334: *Aug 12 18:44:38.117: //24875/99847ABD9C8B/SIP/Error/sipSPI_ipip_GetPassthruContent:
Content-Type: Not present in SIP Message
013335: *Aug 12 18:44:38.121: //-1/99847ABD9C8B/SIP/Error/sip_iwf_def_sdp_pthru_err_disconnect_hdlr:
SDP passthru not supported for non SIP-SIP call, disconnecting call
013336: *Aug 12 18:44:38.121: //-1/xxxxxxxxxxxx/SIP/Error/sipSPIGetContentQSIG:
No Inbound Container Created !!!
013337: *Aug 12 18:44:38.121: //-1/xxxxxxxxxxxx/SIP/Error/sipSPIGetContentQ931:
No Inbound Container Created !!!
013338: *Aug 12 18:44:38.121: //-1/99847ABD9C8B/SIP/Error/sipSPI_ipip_ExtractAndAddPassthruCopyListDataToContainer:
Container is NULL
013339: *Aug 12 18:44:38.121: //-1/xxxxxxxxxxxx/SIP/Error/ccsip_iwf_process_event:
Dead CCB
013340: *Aug 12 18:44:38.129: //-1/99847ABD9C8B/SIP/Error/sipSPIFlushDeferredQueue:
Invalid deferredQueue
013341: *Aug 12 18:44:38.129: //-1/xxxxxxxxxxxx/SIP/Error/sipSPI_ipip_FreeSipRawSdp:
no sdp to free

 

 

HI friends,

 

The translation didnt work. In fact I did not detail the scenario as it should. 

 

- As you can see on topology, the network for sip is 10.7.75.152/29, so on my router i configured the IP: 10.7.75.154, and router of ISP is 10.7.75.153. But my network of VoIP (CUCM and GW voice) is 192.168.0.0/24. So to communication beetween my network and ISP VoIP, it is necessary NAT. I configured the NAT on my GW gateway (192.168.0.100). Follow the topology with details, and show run from gw , attached:

 

tac.jpg

 

 

Follow the outputs requested:

 

debug ccsip messages:

 

013342: *Aug 12 21:58:38.572: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:11307431111@192.168.0.100 SIP/2.0
Max-Forwards: 63
Supported: 100rel
To: <sip:+5511307431111@10.255.241.68>
From: <sip:+5511976637000@10.255.241.68>;tag=3743101251-1978254901
P-Asserted-Identity: <sip:976637000@ims4.vivo.net.br>
Call-ID: 12154930-3743101251-359155644@ngn-br-spo-be-sbc1.mydomain.com
CSeq: 1 INVITE
Allow: PUBLISH,MESSAGE,UPDATE,PRACK,SUBSCRIBE,REFER,INFO,NOTIFY,REGISTER,OPTIONS,BYE,INVITE,ACK,CANCEL
Via: SIP/2.0/UDP 10.255.241.68:5060;branch=z9hG4bK70de28b3e9a8d8aef3194d24cf666029
Contact: <sip:+5511976637000@10.255.241.68:5060>
Content-Type: application/sdp
Accept: application/sdp
P-Charging-Vector: icid-value=mgcf--20180812192051-100023867;orig-ioi=241;term-ioi=3239
P-Notification: caller-control
Content-Length: 216

v=0
o=ngn-br-spo-be-sbc1 201975726 201975726 IN IP4 10.255.241.68
s=sip call
c=IN IP4 10.255.241.76
t=0 0
m=audio 38322 RTP/AVP 8 100
a=rtpmap:100 telephone-event/8000
a=fmtp:100 0-15
a=ptime:20
a=sendrecv

013343: *Aug 12 21:58:38.584: //25068/B3C6CF8B9D52/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.255.241.68:5060;branch=z9hG4bK70de28b3e9a8d8aef3194d24cf666029
From: <sip:+5511976637000@10.255.241.68>;tag=3743101251-1978254901
To: <sip:+5511307431111@10.255.241.68>
Date: Sun, 12 Aug 2018 21:58:38 GMT
Call-ID: 12154930-3743101251-359155644@ngn-br-spo-be-sbc1.mydomain.com
CSeq: 1 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-15.5.3.M
Content-Length: 0


013344: *Aug 12 21:58:38.584: //25069/B3C6CF8B9D52/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:11307431111@192.168.0.103:5060 SIP/2.0
Via: SIP/2.0/UDP 10.7.75.154:5060;branch=z9hG4bK1DEBA0
From: <sip:976637000@10.7.75.154>;tag=F8C460E4-2F8
To: <sip:11307431111@192.168.0.103>
Date: Sun, 12 Aug 2018 21:58:38 GMT
Call-ID: B3C807DB-9DB111E8-9D58BC10-BA50BE76@10.7.75.154
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 3016150923-2645627368-2639445008-3125853814
User-Agent: Cisco-SIPGateway/IOS-15.5.3.M
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1534111118
Contact: <sip:976637000@10.7.75.154:5060>
Call-Info: <sip:10.7.75.154:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
Expires: 180
Allow-Events: kpml, telephone-event
Max-Forwards: 62
P-Asserted-Identity: <sip:976637000@10.7.75.154>
P-Charging-Vector: icid-value=mgcf--20180812192051-100023867;orig-ioi=241;term-ioi=3239
P-Notification: caller-control
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 214

v=0
o=ngn-br-spo-be-sbc1 201975726 201975726 IN IP4 10.255.241.68
s=sip call
c=IN IP4 10.7.75.154
t=0 0
m=audio 18638 RTP/AVP 8 100
a=rtpmap:100 telephone-event/8000
a=fmtp:100 0-15
a=ptime:20
a=sendrecv

013345: *Aug 12 21:58:38.584: //25069/B3C6CF8B9D52/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.7.75.154:5060;branch=z9hG4bK1DEBA0
From: <sip:976637000@10.7.75.154>;tag=F8C460E4-2F8
To: <sip:11307431111@192.168.0.103>
Date: Sun, 12 Aug 2018 22:20:51 GMT
Call-ID: B3C807DB-9DB111E8-9D58BC10-BA50BE76@10.7.75.154
CSeq: 101 INVITE
Allow-Events: presence
Content-Length: 0


013346: *Aug 12 21:58:38.588: //25069/B3C6CF8B9D52/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 10.7.75.154:5060;branch=z9hG4bK1DEBA0
From: <sip:976637000@10.7.75.154>;tag=F8C460E4-2F8
To: <sip:11307431111@192.168.0.103>;tag=1407918946
Date: Sun, 12 Aug 2018 22:20:51 GMT
Call-ID: B3C807DB-9DB111E8-9D58BC10-BA50BE76@10.7.75.154
CSeq: 101 INVITE
Allow-Events: presence
Warning: 399 CUCM "Unable to find a device handler for the request received on port 51614 from 10.7.75.154"
Content-Length: 0


013347: *Aug 12 21:58:38.592: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:11307431111@192.168.0.103:5060 SIP/2.0
Via: SIP/2.0/UDP 10.7.75.154:5060;branch=z9hG4bK1DEBA0
From: <sip:976637000@10.7.75.154>;tag=F8C460E4-2F8
To: <sip:11307431111@192.168.0.103>;tag=1407918946
Date: Sun, 12 Aug 2018 21:58:38 GMT
Call-ID: B3C807DB-9DB111E8-9D58BC10-BA50BE76@10.7.75.154
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: kpml, telephone-event
Content-Length: 0


013348: *Aug 12 21:58:38.592: //25068/B3C6CF8B9D52/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 500 Internal Server Error
Via: SIP/2.0/UDP 10.255.241.68:5060;branch=z9hG4bK70de28b3e9a8d8aef3194d24cf666029
From: <sip:+5511976637000@10.255.241.68>;tag=3743101251-1978254901
To: <sip:+5511307431111@10.255.241.68>;tag=F8C460EC-1ABB
Call-ID: 12154930-3743101251-359155644@ngn-br-spo-be-sbc1.mydomain.com
CSeq: 1 INVITE
Content-Length: 0


013349: *Aug 12 21:58:38.600: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:11307431111@192.168.0.100 SIP/2.0
Max-Forwards: 70
To: <sip:+5511307431111@10.255.241.68>;tag=F8C460EC-1ABB
From: <sip:+5511976637000@10.255.241.68>;tag=3743101251-1978254901
Call-ID: 12154930-3743101251-359155644@ngn-br-spo-be-sbc1.mydomain.com
CSeq: 1 ACK
Via: SIP/2.0/UDP 10.255.241.68:5060;branch=z9hG4bK70de28b3e9a8d8aef3194d24cf666029
Contact: <sip:+5511976637000@10.255.241.68:5060>
Content-Length: 0

 

debug ccsip error:

 

013352: *Aug 12 22:01:46.024: //-1/xxxxxxxxxxxx/SIP/Error/httpish_msg_free:
Freeing NULL pointer!
013353: *Aug 12 22:01:46.024: //25074/2381B7A59D5D/SIP/Error/sipSPIGetPeerByCalledPartyId:
input arg error
013354: *Aug 12 22:01:46.024: //25074/2381B7A59D5D/SIP/Error/sipSPIUpdateCallInfo:
input argument error
013355: *Aug 12 22:01:46.024: //25074/2381B7A59D5D/SIP/Error/ccsip_ipip_media_forking_anchor_leg_config:
MF: Dial-peer is absent..
013356: *Aug 12 22:01:46.024: //25074/2381B7A59D5D/SIP/Error/ccsip_ipip_media_forking_intra_frame_request_config:
MF:video profile Dial-peer is absent..
013357: *Aug 12 22:01:46.024: //25074/2381B7A59D5D/SIP/Error/sipSPI_ipip_build_consolidated_header_list:
No headers associated with passthrulist tag: 0 and copylist tag: 0
013358: *Aug 12 22:01:46.024: //25074/2381B7A59D5D/SIP/Error/sipSPIFlushDeferredQueue:
Invalid deferredQueue
013359: *Aug 12 22:01:47.732: //-1/xxxxxxxxxxxx/SIP/Error/httpish_msg_free:
Freeing NULL pointer!
013360: *Aug 12 22:01:47.732: //-1/248655459D5E/SIP/Error/sipSPI_ipip_build_consolidated_header_list:
No headers associated with passthrulist tag: 0 and copylist tag: 0
SIP: Group (a= group line) attribute, level 65535 instance 1 not found.
SIP: Attribute mid, level 1 instance 1 not found.
013361: *Aug 12 22:01:47.736: //25076/248655459D5E/SIP/Error/sipSPIGetCallServerGroupTargets:
No server group configured
013362: *Aug 12 22:01:47.736: //25076/248655459D5E/SIP/Error/sipSPI_ipip_build_consolidated_header_list:
No headers associated with passthrulist tag: 0 and copylist tag: 0
013363: *Aug 12 22:01:47.744: //25076/248655459D5E/SIP/Error/sipSPI_ipip_GetPassthruContent:
Content-Type: Not present in SIP Message
013364: *Aug 12 22:01:47.744: //-1/248655459D5E/SIP/Error/sip_iwf_def_sdp_pthru_err_disconnect_hdlr:
SDP passthru not supported for non SIP-SIP call, disconnecting call
013365: *Aug 12 22:01:47.744: //-1/xxxxxxxxxxxx/SIP/Error/sipSPIGetContentQSIG:
No Inbound Container Created !!!
013366: *Aug 12 22:01:47.744: //-1/xxxxxxxxxxxx/SIP/Error/sipSPIGetContentQ931:
No Inbound Container Created !!!
013367: *Aug 12 22:01:47.744: //-1/248655459D5E/SIP/Error/sipSPI_ipip_ExtractAndAddPassthruCopyListDataToContainer:
Container is NULL
013368: *Aug 12 22:01:47.744: //-1/xxxxxxxxxxxx/SIP/Error/ccsip_iwf_process_event:
Dead CCB
013369: *Aug 12 22:01:47.756: //-1/248655459D5E/SIP/Error/sipSPIFlushDeferredQueue:
Invalid deferredQueue
013370: *Aug 12 22:01:47.756: //-1/xxxxxxxxxxxx/SIP/Error/sipSPI_ipip_FreeSipRawSdp:
no sdp to free

 

The CUCM system at 192.168.0.103 is denying the SIP invite from the gateway at 10.7.75.154. This is likely due to the voice gateway routing the SIP invite from the wrong IP of the router, and CUCM doesn't recognize the IP. To get to the reason for this, we'll need to see the CUCM dial-peer configs. Could you provide the dial-peer config for CUCM? Also, could you provide the ip routes configured on the gateway? This could impact IPs like this also.

 

013346: *Aug 12 21:58:38.588: //25069/B3C6CF8B9D52/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 10.7.75.154:5060;branch=z9hG4bK1DEBA0
From: <sip:976637000@10.7.75.154>;tag=F8C460E4-2F8
To: <sip:11307431111@192.168.0.103>;tag=1407918946
Date: Sun, 12 Aug 2018 22:20:51 GMT
Call-ID: B3C807DB-9DB111E8-9D58BC10-BA50BE76@10.7.75.154
CSeq: 101 INVITE
Allow-Events: presence
Warning: 399 CUCM "Unable to find a device handler for the request received on port 51614 from 10.7.75.154"
Content-Length: 0

 

 

Ryan

HI sorrry for delay, after apply on dial-peer with  ISP SIP the follow command, all worked:

 

dial-peer voice 1002 voip

description SIP ISP
voice-class sip rel1xx disable
voice-class sip bind control source-interface BVI1
voice-class sip bind media source-interface BVI1
codec g711ulaw

 

dial-peer voice 2 voip
description CALLs to CUCM
voice-class sip rel1xx disable
codec g711ulaw

 

TKs

 

 

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