Hi, I'm having terrible trouble getting a Tecom IP2005 Sip phone working with Asterisk 1.0
I installed Asterisk couple weeks ago, then installed a X100P card and tested with X-Link softphone, all seemed well. So I thought I would buy a Sip phone from a UK company. However I cannot seem to get it to authorise with Asterisk. This is a link to the mfcr website :- http://www.tecomproduct.com/IP2005.htm And a link to the UK suppliers site:- http://www.solwise.co.uk/voip-phones-ip2005.htm Now with sip debug on I see messages like this: Sip read: REGISTER sip:192.168.1.2 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.245:5060;branch=z9hG4bKcKaaZI5Ft Max-Forwards: 70 User-Agent: Centrality PA1688 From: home <sip:[EMAIL PROTECTED]>;tag=yoyzIb5v2ZNzx08i To: home <sip:[EMAIL PROTECTED]> Call-ID: kv3Hc37gOQL6pI4k CSeq: 17455 REGISTER Contact: <sip:[EMAIL PROTECTED]:5060> Expires: 360 Content-Length: 0 11 headers, 0 lines Using latest request as basis request Sending to 192.168.1.245 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.245:5060;branch=z9hG4bKcKaaZI5Ft From: home <sip:[EMAIL PROTECTED]>;tag=yoyzIb5v2ZNzx08i To: home <sip:[EMAIL PROTECTED]>;tag=as249efa19 Call-ID: kv3Hc37gOQL6pI4k CSeq: 17455 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:[EMAIL PROTECTED]> Content-Length: 0 to 192.168.1.245:5060 Transmitting (no NAT): SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.245:5060;branch=z9hG4bKcKaaZI5Ft From: home <sip:[EMAIL PROTECTED]>;tag=yoyzIb5v2ZNzx08i To: home <sip:[EMAIL PROTECTED]>;tag=as249efa19 Call-ID: kv3Hc37gOQL6pI4k CSeq: 17455 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:[EMAIL PROTECTED]> WWW-Authenticate: Digest realm="asterisk", nonce="4532aca5" Content-Length: 0 to 192.168.1.245:5060 Scheduling destruction of call 'kv3Hc37gOQL6pI4k' in 15000 ms splat*CLI> The phone itself just displays "Failed login" message. The phone did come with some firmware which is supposed to give it SIP functionality, I've loaded this on and configured the sip server 192.168.1.2 in the phone. The phone IP is 192.168.1.245. Here is the section from sip.conf [home] type=friend username=home secret=secret callerid="home1" <14> ;host=dynamic port=5060 defaultip=192.168.1.245 nat=no dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info canreinvite=yes ; Typically set to NO if behind NAT disallow=all allow=g723.1 allow=gsm ; GSM consumes far less bandwidth than ulaw allow=ulaw allow=alaw context=sip mailbox=2002 I'd appreciate any help! Many thanks, Mike _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
