Hello, I am using a SIP account from Netvoip CH. I try to receive inbound call from this SIP trunk. I discovered that, when they sent an invite, the IP-Adress of the to: is their own IP address. There fore ACL doesn't work and FS asks for authorization, which then fails
I receive the following message on the CLI: 2009-04-02 21:48:20 [WARNING] sofia_reg.c:1661 sofia_reg_parse_auth() Can't find user [[email protected]] You must define a domain called '62.65.128.62' in your directory and add a user with the id="anonymous" attribute and you must configure your device to use the proper domain in it's authentication credentials. I could do that, but this is not clean and I do not have a password for that. How can I workaround this, so that Freeswitch accepts this call? Aliases do not seem to work. Here is a sample message after FS asks for authorization: xx.xx.xxx.xxx is the IP of our Freeswitch 62.65.128.62 is the IP of Netvoip CH I would expect To: <sip:[email protected]>. instead of To: <sip:[email protected]>. U 62.65.128.62:5060 -> xx.xx.xxx.xxx:5080 INVITE sip:[email protected]:5080 SIP/2.0. Via: SIP/2.0/UDP 62.65.128.62:5060. Via: SIP/2.0/UDP 62.65.128.61:5060;branch=z9hG4bK8c977d2613c4d7d1fd9d03d4. Max-Forwards: 69. From: <sip:[email protected]>;tag=8c977d2613672832fd9d03e9. To: <sip:[email protected]>. Call-ID: [email protected]. CSeq: 2 INVITE. User-agent: Netstream VoIP Gateway. Contact: <sip:[email protected]:5060>. Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,REFER,SUBSCRIBE. Content-Type: application/sdp. Content-Length: 584. Proxy-Authorization: Digest username="anonymous", realm="62.65.128.62", nonce="a4151ee0-1fbb-11de-b056-494b9de21e06", nc="00000001", uri="sip:[email protected]:5060", cnonce="5f109eee", response="62faa6d38b3b12c3626753395a8b507c", algorithm="MD5", qop="auth". . v=0. o=- 225947743692042 1 IN IP4 62.65.128.62. s=-. c=IN IP4 62.65.128.62. t=0 0. m=audio 28224 RTP/AVP 8 18 4 3 100 100 99 100 100 98 97 96 105 0 101. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:4 G723/8000. a=fmtp:4 annexa=no. a=rtpmap:3 GSM/8000. a=rtpmap:100 speex/8000. a=rtpmap:100 speex/8000. a=rtpmap:99 G726-16/8000. a=rtpmap:100 speex/8000. a=rtpmap:100 speex/8000. a=rtpmap:98 G726-24/8000. a=rtpmap:97 G726-32/8000. a=rtpmap:96 G726-40/8000. a=rtpmap:105 iLBC/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. Best regards Peter _______________________________________________ Freeswitch-users mailing list [email protected] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
