05-20-2014 01:47 PM - edited 03-16-2019 10:50 PM
Hello Experts,
I have an requirement to configure a SIP trunk to one of the cloud provider who has a Asterisk PBX. Requirement is to register the SIP trunk on router with the cloud pbx. Then the calling number assigned should be able to route to an extension on a different call manager.
PSTN -> SIP Router -> Call Manager -> Extension
Appreciate if anybody can suggest suitable configuration required to make it work.
Thanks
Arvind
Solved! Go to Solution.
05-21-2014 12:45 AM
First you need to enable CUBE features on your 2900 router. Then configure inbound , outbound SIP dial-peer for Cloud PBX and CUCM. Create a SIP trunk with correct CSS on your CUCM to handle inbound/outbound call from extensions to Cloud PBX.
If your Cloud PBX provider required authentication then you need to configure authentication under sip-ua on 2900 router. Go through these below link it will help you to configure base configuration and if you face any problem get back to this forum we will help you to get it corrected.
https://supportforums.cisco.com/discussion/11765626/cube-configuration-cucm-call-manager
https://supportforums.cisco.com/document/69976/frequently-asked-questions-cisco-unified-border-element-cube
http://www.cisco.com/c/en/us/td/docs/ios/12_2/voice/configuration/guide/fvvfax_c/vvfsip.html
Thanks
Manish
05-21-2014 12:45 AM
First you need to enable CUBE features on your 2900 router. Then configure inbound , outbound SIP dial-peer for Cloud PBX and CUCM. Create a SIP trunk with correct CSS on your CUCM to handle inbound/outbound call from extensions to Cloud PBX.
If your Cloud PBX provider required authentication then you need to configure authentication under sip-ua on 2900 router. Go through these below link it will help you to configure base configuration and if you face any problem get back to this forum we will help you to get it corrected.
https://supportforums.cisco.com/discussion/11765626/cube-configuration-cucm-call-manager
https://supportforums.cisco.com/document/69976/frequently-asked-questions-cisco-unified-border-element-cube
http://www.cisco.com/c/en/us/td/docs/ios/12_2/voice/configuration/guide/fvvfax_c/vvfsip.html
Thanks
Manish
05-30-2014 03:19 PM
05-30-2014 03:51 PM
Did you collected the debugs for the failed call. if not can you please do so.
Make a call collect following debugs
PSTN -----SIP ---- SIP Router ---SIP/H323 Call Manager -> Extension
1) debug ccsip messages
2) debug voice ccapi inout
3) Debug h225 asn1 and debug h245 asn1(if connection from router to Call manager is h.323)
br,
nadeem
05-30-2014 05:55 PM
Thanks for the response Nadeem,
Here is the partial output. the router got hung, waiting for it to reboot.
May 31 00:37:01.623: //34376/7EE343D99365/CCAPI/cc_api_call_disconnected:
Cause Value=16, Interface=0x314B1908, Call Id=34376
May 31 00:37:01.623: //34376/7EE343D99365/CCAPI/cc_api_call_disconnected:
Call Entry(Responsed=FALSE, Cause Value=16, Retry Count=0)
May 31 00:37:01.623: //34377/7EE343D99367/CCAPI/cc_api_call_disconnected:
Cause Value=16, Interface=0x314B1908, Call Id=34377
May 31 00:37:01.623: //34377/7EE343D99367/CCAPI/cc_api_call_disconnected:
Call Entry(Responsed=FALSE, Cause Value=16, Retry Count=0)
May 31 00:37:01.623: //34378/7EE343D99369/CCAPI/cc_api_call_disconnected:
Cause Value=16, Interface=0x314B1908, Call Id=34378
May 31 00:37:01.623: //34378/7EE343D99369/CCAPI/cc_api_call_disconnected:
Call Entry(Responsed=FALSE, Cause Value=16, Retry Count=0)
May 31 00:37:01.623: //34379/7EE343D9936B/CCAPI/cc_api_call_disconnected:
Cause Value=16, Interface=0x314B1908, Call Id=34379
May 31 00:37:01.623: //34379/7EE343D9936B/CCAPI/cc_api_call_disconnected:
Call Entry(Responsed=FALSE, Cause Value=16, Retry Count=0)
May 31 00:37:01.623: //34380/7EE343D9936D/CCAPI/cc_api_call_disconnected:
Cause Value=16, Interface=0x314B1908, Call Id=34380
May 31 00:37:01.623: //34380/7EE343D9936D/CCAPI/cc_api_call_disconnected:
05-30-2014 05:58 PM
06-02-2014 06:09 AM
Hello Arvind,
I have checked the shared logs unfortunately it didn't give any idea for the call failure. however seems like there is no calling and called party information being received from incoming side and on SIP router its hitting the default Dial-peer ie 0.
May 31 00:37:03.223: //-1/84E5F5EA93AF/CCAPI/cc_api_call_setup_ind_common:
Interface=0x314B1908, Call Info(
Calling Number=,(Calling Name=)(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed),
Called Number=(TON=Unknown, NPI=Unknown),
Calling Translated=FALSE, Subscriber Type Str=RegularLine, FinalDestinationFlag=FALSE,
Incoming Dial-peer=0, Progress Indication=ORIGINATING SIDE IS NON ISDN(3), Calling IE Present=FALSE,
Source Trkgrp Route Label=, Target Trkgrp Route Label=, CLID Transparent=FALSE), Call Id=-1
May 31 00:37:03.223: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
May 31 00:37:03.223: :cc_get_feature_vsa malloc success
May 31 00:37:03.223: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
May 31 00:37:03.223: cc_get_feature_vsa count is 3
May 31 00:37:03.223: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
May 31 00:37:03.223: :FEATURE_VSA attributes are: feature_name:0,feature_time:833283240,feature_id:34184
May 31 00:37:03.223: //34389/84E5F5EA93AF/CCAPI/cc_api_call_setup_ind_common:
Set Up Event Sent;
Call Info(Calling Number=(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed),
Called Number=(TON=Unknown, NPI=Unknown))
May 31 00:37:03.223: //-1/84E5F5EA93B1/CCAPI/cc_api_display_ie_subfields:
cc_api_call_setup_ind_common:
cisco-username=
----- ccCallInfo IE subfields -----
cisco-ani=
cisco-anitype=0
cisco-aniplan=0
cisco-anipi=0
cisco-anisi=0
dest=
cisco-desttype=0
cisco-destplan=0
cisco-rdie=FFFFFFFF
cisco-rdn=
cisco-rdntype=0
cisco-rdnplan=0
cisco-rdnpi=0
cisco-rdnsi=0
cisco-redirectreason=0 fwd_final_type =0
final_redirectNumber =
hunt_group_timeout =0
could you please re-collect the logs with said debugs and also please mention the calling and called party number.
use this link to collect the logs on router safely https://supportforums.cisco.com/document/62906/how-properly-and-safely-collect-debugs-ios-routeror would advise collect in low traffic time.
06-22-2024 06:31 AM
Hello There...
I have cisco 2900 series router and we transformed to sip trunk recentrly, does this router model supports sip? needs a special license?
and how's the configurations please.
My network design is like this PSTN>>>Voice GW Router 2900>>>FIREWALL>>>Call Manager
06-22-2024 06:51 AM
A few things, first off your question is off topic to the OP, secondly the post is set as Solved, so adding questions to it is not considered best practice, thirdly the post is ten years old. You’ll be better off by creating your own post to ask your question.
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