[Asterisk-Users] Cisco Call Manager SIP trunk + Asterisk

[Chris HARIGA]

Hi,

 

I setup a SIP trunk in a brand new Cisco Call Manager and I try to place the
calls using Asterisk. but I get error:

 

"<-- SIP read from 192.168.11.10:5060:

SIP/2.0 400 Bad Request - 'Malformed/Missing URL'

Via: SIP/2.0/UDP 192.168.10.199:5060;branch=z9hG4bK2e7ca9c9;rport

From: "asterisk" <sip:asterisk at 192.168.10.199>;tag=as56c7728f

To: <sip:192.168.11.10>

Call-ID: 299a873b30ad20f90bbcb66e3d505e68 at 192.168.10.199

CSeq: 102 OPTIONS

Content-Length: 0"

 

Question: How I can setup asterisk to get the sip call without
authentication? I check on voip-info.org but I didn't find a sip.conf sample
:-(

 

Best regards,

 

Chris HARIGA

 

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