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SIP Trunk configuration onCisco 2900 router for inbound calling

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

1 Accepted Solution

Accepted Solutions

Manish Prasad
Level 5
Level 5

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

View solution in original post

8 Replies 8

Manish Prasad
Level 5
Level 5

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

Hi Manish,

Now we getting incoming calls when we dial international from India, but does not works when we dial locally from USA. Any thoughts?

 

Attached is the configuration.

 

Thanks

Arabinda

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

Br, Nadeem Please rate all useful post.

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:

 

uploading the log file, appreciate your findings on this.

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.

 

 

 

Br, Nadeem Please rate all useful post.

kardosabr
Level 1
Level 1

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

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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