asterisk users - Jan 2005 - Re: Asterisk-Users Digest, Vol 6, Issue 105

Scott, Right now I am using the Netphone KE1020A. ~Dan Message: 11 Date: 
Sat, 08 Jan 2005 13:34:29 -0900 From: Scott Henderson 
<scott@finite-tech.com> Subject: Re: [Asterisk-Users] Re: Connecting Sip 
phone to asterisk. To: Asterisk Users Mailing List - Non-Commercial 
Discussion <asterisk-users@lists.digium.com> Message-ID: 
<41E05FF5.30302@finite-tech.com> Content-Type: text/plain; 
charset=ISO-8859-1; format=flowed One more thing, what phones are you 
using? joosfamily@speakeasy.net wrote:
>>The phone is configured as:
>>IP Phone Number: 1201
>>Username: 1201
>>Password: <password>
>>Service Address: 192.168.0.104
>>
>>Sip.conf is configured as:
>>[1201]
>>type=friend
>>username=1201
>>secret=<password>
>>mailbox=1201
>>host=192.168.0.99
>>
>>
>>To keep the redundant data down, here is what the sip debug shows:
>>
>>Retransmitting #5 (no NAT):
>>INVITE sip:1201@192.168.0.99 SIP/2.0
>>Via: SIP/2.0/UDP 192.168.0.104:5060;branch=z9hG4bK6a747077
>>From: "1202" <sip:1202@192.168.0.104>;tag=as166994fa
>>To: <sip:1201@192.168.0.99>
>>Contact: <sip:1202@192.168.0.104>
>>Call-ID: 5ac70b3e78833fd40669cb867ba3ccb7@192.168.0.104
>>CSeq: 102 INVITE
>>User-Agent: Asterisk PBX
>>Date: Sat, 08 Jan 2005 20:22:19 GMT
>>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
>>Content-Type: application/sdp
>>Content-Length: 263
>>
>>v=0
>>o=root 6552 6552 IN IP4 192.168.0.104
>>s=session
>>c=IN IP4 192.168.0.104
>>t=0 0
>>m=audio 11670 RTP/AVP 0 3 8 101
>>a=rtpmap:0 PCMU/8000
>>a=rtpmap:3 GSM/8000
>>a=rtpmap:8 PCMA/8000
>>a=rtpmap:101 telephone-event/8000
>>a=fmtp:101 0-16
>>a=silenceSupp:off - - - -
>>
>> to 192.168.0.99:5060
>>Jan  8 12:22:25 WARNING[6552]: chan_sip.c:694 retrans_pkt: Maximum
retries exceeded on call 5ac70b3e78833fd40669cb867ba3ccb7@192.168.0.104 for
seqno 102 (Critical Request)
>>Destroying call '5ac70b3e78833fd40669cb867ba3ccb7@192.168.0.104'
>>
>>
>>_______________________________________________
>
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