Greetings, we have an existing UC500 running v12.4.22YB8 (CME v7.1) and need to create a SIP trunk for inter-office calling. The remote office is running Asterisk (Grandstream 126.96.36.199). The UC500 is the firewall and the Internet gateway at the main office. There is a VPN created between the UC500 and the remote office with the voice LAN and remote LAN NAT exempted. The UC500 can ping the Asterisk from its voice VLAN. UC500 IP is 10.1.1.1, Asterisk IP is 192.168.42.40 and username for authentication of the SIP trunk is 3611
Astrisk shows the trunk setup is rejected. Debugs on the UC500 show no match.
Debug ccsip error and messages shows the following:
321131: Aug 11 17:53:51.764 AST: //-1/xxxxxxxxxxxx/SIP/Error/ccsip_reg_search_call_id_table: No entry found in sipCallID Table 321132: Aug 11 17:53:51.764 AST: //-1/xxxxxxxxxxxx/SIP/Error/sipSPI_validate_own_ip_addr: ReqLine IP addr does not match with host IP addr 321133: Aug 11 17:53:51.768 AST: //388436/D0E4D95D99A7/SIP/Error/ccsip_spi_register_incoming_registration: No matching pool for this REGISTER
Thanks for the reply. I did see these posts and quite a few more while searching, but all of these are referencing phones attaching to the CME. I am kind of confused. This is regarding setting up a SIP trunk between CME and Asterisk. I've done many of these with CME to CME or CM without needing voice register pools. Also, all of the configuration examples for SIP trunks on CME don't mention anything about voice register pools. Does a SIP trunk from a non Cisco PBX work different than Cisco to Cisco but instead work as a phone?
The reason I sent you those links is because your asterisk is trying to register to the CCME. I am not sure why you need an active registration. If you dont need one, you should just configure a sip dial-peer to asterisk and that should work
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