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Created by
Thiago Cella
in IP Telephony and Phones
in IP Telephony and Phones
09-06-2018
03:51 PM
09-06-2018
03:51 PM
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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Created by
Thiago Cella
in IP Telephony and Phones
in IP Telephony and Phones
08-12-2018
03:25 PM
08-12-2018
03:25 PM
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:
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
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Created by
Thiago Cella
in IP Telephony and Phones
in IP Telephony and Phones
08-12-2018
12:11 PM
08-12-2018
12:11 PM
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;
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
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Created by
Thiago Cella
in IP Telephony and Phones
in IP Telephony and Phones
08-10-2018
06:47 PM
08-10-2018
06:47 PM
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
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08-10-2018
05:30 AM
Hi, follow the fields to change, Tks :
INVITE sip:206832XX@10.7.75.154:5060 SIP/2.0 to INVITE sip:206832XX@10.255.240.111:5060 SIP/2.0
Contact URI :sip:399@10.255.240.111:5060 to Contact URI: sip:11307437XX@10.7.75.154:5060
From: "Suporte" <sip:399@10.7.75.154> to From: <sip:11307437XX@10.255.240.111>
Request URI :sip:399@10.255.240.111:5060 to Request URI : sip:11307437XX@10.7.75.154:5060
Tks
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08-09-2018
04:42 PM
Hi Friends,
I need help to configure the sip profile , with the follow parameters. Could you help me?
Message Header Via: SIP/2.0/UDP 10.7.75.154:5060;branch=z9hG4bK-524287-1---76e9c445195aa645;rport Max-Forwards: 70 Contact: <sip:11307437XX@10.7.75.154:5060> Contact URI: sip:11307437XX@10.7.75.154:5060 To: <sip:206832XX@10.255.240.111> SIP to address: sip:206832XX@10.255.240.111 From: <sip:11307437XX@10.255.240.111>;tag=dd48107b SIP from address: sip:11307437XX@10.255.240.111 SIP from tag: dd48107b Call-ID: 92990NGMyOWYzNDlhZDQ3MjcyYzZlMzc1M2IxYTViNjY3MjQ CSeq: 1 INVITE Sequence Number: 1 Method: INVITE Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, OPTIONS, MESSAGE Content-Type: application/sdp Supported: replaces User-Agent: X-Lite release 5.3.3 stamp 92990 Content-Length: 332 Message Body
The follow that is current header when i make a test, but it is wrong, the call dont work:
INVITE sip:206832XX@10.255.240.111:5060 SIP/2.0 Via: SIP/2.0/UDP 10.7.75.154:5060;branch=z9hG4bKDC834 From: "Suporte" <sip:399@10.7.75.154>;tag=E996948C-112A To: <sip:206832XX@10.255.240.111> Date: Thu, 09 Aug 2018 23:14:18 GMT Call-ID: C6CC873A-9B6011E8-8599BC10-BA50BE76@10.7.75.154 Supported: timer,resource-priority,replaces,sdp-anat Min-SE: 1800 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: 1533856458 Contact: <sip:399@10.7.75.154:5060> Expires: 180 Allow-Events: telephone-event Max-Forwards: 68 P-Asserted-Identity: "Suporte" <sip:399@10.7.75.154> Session-Expires: 1800 Session-ID: a27bd856b48340c6ed50db2aa5131224;remote=00000000000000000000000000000000 Content-Length: 0 Account-Number:399
TKS
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05-29-2017
05:33 PM
Hi,
Here the same problem with ASA5508, after update the software version to asa981-lfbff-k8.SPA , the problem was fixed.
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08-21-2016
04:13 PM
Hi Friends,
Sorry for the late, the problem was on router of ISP. I made more tests using x-lite and the some calls calls didnt work too. So i configured the link directly on router and all works.
tks
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08-11-2016
05:57 AM
One more information , I made a test with same account with X-lite client on my PC and works.
T he follow LOG is appering on Cisco, when I try make a call. I
Received: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 201.69.228.73:49152;branch=z9hG4bK261412;rport=57056;received=192.168.1.30 From: <sip:1 1111.1111.1111@1.1.1.1>;tag=12E370-1A20 To: <sip:9999999@187.63.130.50>;tag=18f01cbe78aa7d69fd8f3e0a8ea294be.fda4 Call-ID: 6E23CBF1-5EF811E6-802CC533-85BC4FB0@192.168.1.30 CSeq: 101 INVITE Proxy-Authenticate: Digest realm="isp.com", nonce="57ac7536000133f7398173a31fd71115b45acad43b6f9f70" Server: nt Content-Length: 0
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08-10-2016
05:33 PM
Hi
There is no firewall. This test is a outbound Call, i will test inbound Call and report the results.
Tks
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08-10-2016
03:50 PM
Hi,
I configured the follow , but during the call there is no audio on both sides, could you help me?
version 15.1 service timestamps debug datetime msec service timestamps log datetime msec no service password-encryption ! hostname Router ! boot-start-marker boot-end-marker ! ! ! no aaa new-model dot11 syslog ip source-route ! ! ! ! ! ip cef no ipv6 cef multilink bundle-name authenticated ! ! ! ! ! ! voice service voip address-hiding allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip redirect ip2ip h323 emptycapability no telephony-service ccm-compatible sip registrar server ! voice class sip-profiles 100 request INVITE sip-header From modify "<sip:(.*)@1.1.1.1>" "<sip:1111. 1 111 . 1 111 @1.1.1.1>" ! ! ! ! ! voice-card 0 ! crypto pki token default removal timeout 0 ! ! ! ! license udi pid CISCO2801 sn FHK1147F1JU license accept end user agreement ! ! ! ! ! ! ! interface FastEthernet0/0 ip address dhcp duplex auto speed auto ! interface FastEthernet0/1 ip address 192.168.23.1 255.255.255.0 duplex auto speed auto ! ip forward-protocol nd ! ! no ip http server no ip http secure-server ! ! ! ! control-plane ! ! voice-port 0/2/0 ! voice-port 0/2/1 ! ! ! mgcp profile default ! ! ! dial-peer voice 1010 voip description OUTBOUND to ISP destination-pattern .T session protocol sipv2 session target sip-server voice-class sip dtmf-relay force rtp-nte voice-class sip profiles 100 dtmf-relay rtp-nte codec g711alaw clid network-number 1122222222 ! ! sip-ua credentials username 1111. 1 111 . 1 111 password XXXXXXXX realm isp.com authentication username 1111. 1 111 . 1 111 password XXXXXXXX realm isp.com no remote-party-id retry invite 2 retry register 10 timers connect 100 registrar ipv4: 1.1.1.1 expires 1800 sip-server ipv4: 1.1.1.1 host-registrar ! ! telephony-service max-ephones 15 max-dn 15 ip source-address 192.168.23.1 port 2000 max-conferences 4 gain -6 transfer-system full-consult create cnf-files version-stamp Jan 01 2002 00:00:00 ! ! ephone-dn 1 dual-line number 1000 no-reg primary ! ephone 1 description joao-teste mac-address B888.E3E0.A621 button 1:1 ! !
Follow debug :
*Aug 10 22:44:47.707: //85/DE99B42C808F/SIP/Msg/ccsipDisplayMsg: Sent: INVITE sip:99999999@1.1.1.1:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.30:5060;branch=z9hG4bK56B2F From: <sip:1111.1111.1111@1.1.1.1>;tag=8E61D8-485 To: <sip:99999999@1.1.1.1> Date: Wed, 10 Aug 2016 22:44:47 GMT Call-ID: DFED0EAF-5E8211E6-8094B238-902B853E@192.168.1.30 Supported: 100rel,timer,resource-priority,replaces,sdp-anat Min-SE: 1800 Cisco-Guid: 3734615084-1585582566-2156900920-2418771262 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER CSeq: 101 INVITE Max-Forwards: 70 Timestamp: 1470869087 Contact: <sip:1122222222@192.168.1.30:5060> Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Disposition: session;handling=required Content-Length: 271
v=0 o=CiscoSystemsSIP-GW-UserAgent 7869 7950 IN IP4 192.168.1.30 s=SIP Call c=IN IP4 192.168.1.30 t=0 0 m=audio 17740 RTP/AVP 8 101 19 c=IN IP4 192.168.1.30 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:19 CN/8000 a=ptime:20
*Aug 10 22:44:47.723: //85/DE99B42C808F/SIP/Msg/ccsipDisplayMsg: Received: SIP/2.0 100 Trying Via: SIP/2.0/UDP 3.3.3.3:49152;branch=z9hG4bK56B2F;rport=64718;received=192.168.1.30 From: <sip:1111.1111.1111@1.1.1.1>;tag=8E61D8-485 To: <sip:99999999@1.1.1.1> Call-ID: DFED0EAF-5E8211E6-8094B238-902B853E@192.168.1.30 CSeq: 101 INVITE Server: nt Content-Length: 0
*Aug 10 22:44:47.727: //85/DE99B42C808F/SIP/Msg/ccsipDisplayMsg: Received: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 3.3.3.3:49152;branch=z9hG4bK56B2F;rport=64718;received=192.168.1.30 From: <sip:1111.1111.1111@1.1.1.1>;tag=8E61D8-485 To: <sip:99999999@1.1.1.1>;tag=d8ba4cf51f1985334ea8c7911ac42cf4.5f85 Call-ID: DFED0EAF-5E8211E6-8094B238-902B853E@192.168.1.30 CSeq: 101 INVITE Proxy-Authenticate: Digest realm="isp.com", nonce="57abaffb00011b77d2f49ecf6a217052132bc67a5f4535a3" Server: nt Content-Length: 0
*Aug 10 22:44:47.735: //85/DE99B42C808F/SIP/Msg/ccsipDisplayMsg: Sent: ACK sip:99999999@1.1.1.1:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.30:5060;branch=z9hG4bK56B2F From: <sip:1122222222@1.1.1.1>;tag=8E61D8-485 To: <sip:99999999@1.1.1.1>;tag=d8ba4cf51f1985334ea8c7911ac42cf4.5f85 Date: Wed, 10 Aug 2016 22:44:47 GMT Call-ID: DFED0EAF-5E8211E6-8094B238-902B853E@192.168.1.30 Max-Forwards: 70 CSeq: 101 ACK Allow-Events: telephone-event Content-Length: 0
*Aug 10 22:44:47.735: //85/DE99B42C808F/SIP/Msg/ccsipDisplayMsg: Sent: INVITE sip:99999999@1.1.1.1:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.30:5060;branch=z9hG4bK57150B From: <sip:1111.1111.1111@1.1.1.1>;tag=8E61D8-485 To: <sip:99999999@1.1.1.1> Date: Wed, 10 Aug 2016 22:44:47 GMT Call-ID: DFED0EAF-5E8211E6-8094B238-902B853E@192.168.1.30 Supported: 100rel,timer,resource-priority,replaces,sdp-anat Min-SE: 1800 Cisco-Guid: 3734615084-1585582566-2156900920-2418771262 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER CSeq: 102 INVITE Max-Forwards: 70 Timestamp: 1470869087 Contact: <sip:1122222222@192.168.1.30:5060> Expires: 180 Allow-Events: telephone-event Proxy-Authorization: Digest username="1111.1111.1111",realm="isp.com",uri="sip:99999999@1.1.1.1:5060",response="aa74f1052b7797de81f57343cd025366",nonce="57abaffb00011b77d2f49ecf6a217052132bc67a5f4535a3",algorithm=md5 Content-Type: application/sdp Content-Disposition: session;handling=required Content-Length: 271
v=0 o=CiscoSystemsSIP-GW-UserAgent 7869 7950 IN IP4 192.168.1.30 s=SIP Call c=IN IP4 192.168.1.30 t=0 0 m=audio 17740 RTP/AVP 8 101 19 c=IN IP4 192.168.1.30 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:19 CN/8000 a=ptime:20
*Aug 10 22:44:47.743: //85/DE99B42C808F/SIP/Msg/ccsipDisplayMsg: Received: SIP/2.0 100 Trying Via: SIP/2.0/UDP 3.3.3.3:49152;branch=z9hG4bK57150B;rport=64718;received=192.168.1.30 From: <sip:1111.1111.1111@1.1.1.1>;tag=8E61D8-485 To: <sip:99999999@1.1.1.1> Call-ID: DFED0EAF-5E8211E6-8094B238-902B853E@192.168.1.30 CSeq: 102 INVITE Server: nt Content-Length: 0
*Aug 10 22:44:52.171: //85/DE99B42C808F/SIP/Msg/ccsipDisplayMsg: Received: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 3.3.3.3:49152;rport=64718;received=192.168.1.30;branch=z9hG4bK57150B Record-Route: <sip:1.1.1.1:5060;lr=on;ftag=8E61D8-485;did=ff7.0f54d5d6;vsf=AAAAAAAABAMeAgkABx94CQgDbgcLGR8FAwAEAzAuNTA-> Require: 100rel Contact: <sip:01199999999@172.16.0.1:5060> To: <sip:99999999@1.1.1.1>;tag=ee7c6277 From: <sip:1111.1111.1111@1.1.1.1>;tag=8E61D8-485 Call-ID: DFED0EAF-5E8211E6-8094B238-902B853E@192.168.1.30 CSeq: 102 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, PRACK, INFO RSeq: 1 Content-Length: 0
*Aug 10 22:44:52.179: //85/DE99B42C808F/SIP/Msg/ccsipDisplayMsg: Sent: PRACK sip:01199999999@172.16.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.30:5060;branch=z9hG4bK581FD4 From: <sip:1122222222@1.1.1.1>;tag=8E61D8-485 To: <sip:99999999@1.1.1.1>;tag=ee7c6277 Date: Wed, 10 Aug 2016 22:44:47 GMT Call-ID: DFED0EAF-5E8211E6-8094B238-902B853E@192.168.1.30 CSeq: 103 PRACK RAck: 1 102 INVITE Route: <sip:1.1.1.1:5060;lr=on;ftag=8E61D8-485;did=ff7.0f54d5d6;vsf=AAAAAAAABAMeAgkABx94CQgDbgcLGR8FAwAEAzAuNTA-> Allow-Events: telephone-event Proxy-Authorization: Digest username="1111.1111.1111",realm="isp.com",uri="sip:01199999999@172.16.0.1:5060",response="b8d26f88b22916bdfbc04c211c4b7fee",nonce="57abaffb00011b77d2f49ecf6a217052132bc67a5f4535a3",algorithm=md5 Max-Forwards: 70 Content-Length: 0
*Aug 10 22:44:52.195: //85/DE99B42C808F/SIP/Msg/ccsipDisplayMsg: Received: SIP/2.0 200 OK Via: SIP/2.0/UDP 3.3.3.3:49152;rport=64718;received=192.168.1.30;branch=z9hG4bK581FD4 Contact: <sip:01199999999@172.16.0.1:5060> To: <sip:99999999@1.1.1.1>;tag=ee7c6277 From: <sip:1122222222@1.1.1.1>;tag=8E61D8-485 Call-ID: DFED0EAF-5E8211E6-8094B238-902B853E@192.168.1.30 CSeq: 103 PRACK Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, PRACK, INFO Content-Length: 0
*Aug 10 22:44:52.867: //85/DE99B42C808F/SIP/Msg/ccsipDisplayMsg: Received: SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 3.3.3.3:49152;rport=64718;received=192.168.1.30;branch=z9hG4bK57150B Record-Route: <sip:1.1.1.1:5060;lr=on;ftag=8E61D8-485;did=ff7.0f54d5d6;vsf=AAAAAAAABAMeAgkABx94CQgDbgcLGR8FAwAEAzAuNTA-> Require: 100rel Contact: <sip:01199999999@172.16.0.1:5060> To: <sip:99999999@1.1.1.1>;tag=ee7c6277 From: <sip:1111.1111.1111@1.1.1.1>;tag=8E61D8-485 Call-ID: DFED0EAF-5E8211E6-8094B238-902B853E@192.168.1.30 CSeq: 102 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, PRACK, INFO Content-Type: application/sdp RSeq: 2 Content-Length: 284
v=0 o=MG4000|2.0 15467 15657 IN IP4 172.11.10.82 s=- c=IN IP4 200.236.219.50 t=0 0 m=audio 38748 RTP/AVP 8 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=X-vrzcap:vbd Ver=0 Mode=FaxPr ModemRtpRed=0 a=X-vrzcap:identification bin=CMG1143 Prot=mgcp App=MG
*Aug 10 22:44:52.887: //85/DE99B42C808F/SIP/Msg/ccsipDisplayMsg: Sent: PRACK sip:01199999999@172.16.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.30:5060;branch=z9hG4bK59858 From: <sip:1122222222@1.1.1.1>;tag=8E61D8-485 To: <sip:99999999@1.1.1.1>;tag=ee7c6277 Date: Wed, 10 Aug 2016 22:44:47 GMT Call-ID: DFED0EAF-5E8211E6-8094B238-902B853E@192.168.1.30 CSeq: 104 PRACK RAck: 2 102 INVITE Route: <sip:1.1.1.1:5060;lr=on;ftag=8E61D8-485;did=ff7.0f54d5d6;vsf=AAAAAAAABAMeAgkABx94CQgDbgcLGR8FAwAEAzAuNTA-> Allow-Events: telephone-event Proxy-Authorization: Digest username="1111.1111.1111",realm="isp.com",uri="sip:01199999999@172.16.0.1:5060",response="b8d26f88b22916bdfbc04c211c4b7fee",nonce="57abaffb00011b77d2f49ecf6a217052132bc67a5f4535a3",algorithm=md5 Max-Forwards: 70 Content-Length: 0
*Aug 10 22:44:52.903: //85/DE99B42C808F/SIP/Msg/ccsipDisplayMsg: Received: SIP/2.0 200 OK Via: SIP/2.0/UDP 3.3.3.3:49152;rport=64718;received=192.168.1.30;branch=z9hG4bK59858 Contact: <sip:01199999999@172.16.0.1:5060> To: <sip:99999999@1.1.1.1>;tag=ee7c6277 From: <sip:1122222222@1.1.1.1>;tag=8E61D8-485 Call-ID: DFED0EAF-5E8211E6-8094B238-902B853E@192.168.1.30 CSeq: 104 PRACK Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, PRACK, INFO Content-Length: 0
*Aug 10 22:44:57.367: //85/DE99B42C808F/SIP/Msg/ccsipDisplayMsg: Received: SIP/2.0 200 OK Via: SIP/2.0/UDP 3.3.3.3:49152;rport=64718;received=192.168.1.30;branch=z9hG4bK57150B Record-Route: <sip:1.1.1.1:5060;lr=on;ftag=8E61D8-485;did=ff7.0f54d5d6;vsf=AAAAAAAABAMeAgkABx94CQgDbgcLGR8FAwAEAzAuNTA-> Contact: <sip:01199999999@172.16.0.1:5060> To: <sip:99999999@1.1.1.1>;tag=ee7c6277 From: <sip:1111.1111.1111@1.1.1.1>;tag=8E61D8-485 Call-ID: DFED0EAF-5E8211E6-8094B238-902B853E@192.168.1.30 CSeq: 102 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, PRACK, INFO Content-Length: 0
*Aug 10 22:44:57.379: //85/DE99B42C808F/SIP/Msg/ccsipDisplayMsg: Sent: ACK sip:01199999999@172.16.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.30:5060;branch=z9hG4bK5AEB7 From: <sip:1122222222@1.1.1.1>;tag=8E61D8-485 To: <sip:99999999@1.1.1.1>;tag=ee7c6277 Date: Wed, 10 Aug 2016 22:44:47 GMT Call-ID: DFED0EAF-5E8211E6-8094B238-902B853E@192.168.1.30 Route: <sip:1.1.1.1:5060;lr=on;ftag=8E61D8-485;did=ff7.0f54d5d6;vsf=AAAAAAAABAMeAgkABx94CQgDbgcLGR8FAwAEAzAuNTA-> Max-Forwards: 70 CSeq: 102 ACK Proxy-Authorization: Digest username="1111.1111.1111",realm="isp.com",uri="sip:99999999@1.1.1.1:5060",response="aa74f1052b7797de81f57343cd025366",nonce="57abaffb00011b77d2f49ecf6a217052132bc67a5f4535a3",algorithm=md5 Allow-Events: telephone-event Content-Length: 0
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02-24-2016
03:26 PM
Hi,
I tried apply the firesight license (FS-VMW-2-SW-K9) on version 6, and the follow message appers:
" FireSIGHT licenses are no longer required for Firepower Management Centers Version 6.0 or later "
It is no longer necessary license since version 6?
tks
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Created by
Thiago Cella
in IP Telephony and Phones
09-06-2018
09-06-2018
HI sorrry for delay, after apply on dial-peer with ISP SIP the follow
command, all worked: dial-peer...
Created by
Thiago Cella
in IP Telephony and Phones
08-12-2018
08-12-2018
HI friends, The translation didnt work. In fact I did not detail the
scenario as it should. - As you...
Created by
Thiago Cella
in IP Telephony and Phones
08-12-2018
08-12-2018
Tks Friends, still dont working the translation. I attached the config
of gw, and the topology.Follo...
Created by
Thiago Cella
in IP Telephony and Phones
08-10-2018
08-10-2018
Hi, Im trying implement translation rule, when someone from outside call
551130741111, the cube have...
08-10-2018
Hi, follow the fields to change, Tks : INVITE
sip:206832XX@10.7.75.154:5060 SIP/2.0toINVITE
sip:2068...
08-09-2018
Hi Friends, I need help to configure the sip profile , with the follow
parameters. Could you help me...
05-29-2017
Hi, Here the same problem with ASA5508, after update the software
version to asa981-lfbff-k8.SPA , t...
08-21-2016
Hi Friends, Sorry for the late, the problem was on router of ISP. I made
more tests using x-lite and...
08-11-2016
One more information , I made a test with same account with X-lite
client on my PC and works. The fo...
08-10-2016
Hi There is no firewall. This test is a outbound Call, i will test
inbound Call and report the resul...
08-10-2016
Hi, I configured the follow , but during the call there is no audio on
both sides, could you help me...
Public Statistics
| Date Registered | 08-10-2010 07:58 AM |
| Date Last Visited | 04-01-2020 04:53 AM |
| Total Messages Posted | 172 |
