[Asterisk-Users] Re: Connecting Sip phone to asterisk.

[Scott Henderson]

One more thing, what phones are you using?

joosfamily at 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 at 192.168.0.99 SIP/2.0
>Via: SIP/2.0/UDP 192.168.0.104:5060;branch=z9hG4bK6a747077
>From: "1202" <sip:1202 at 192.168.0.104>;tag=as166994fa
>To: <sip:1201 at 192.168.0.99>
>Contact: <sip:1202 at 192.168.0.104>
>Call-ID: 5ac70b3e78833fd40669cb867ba3ccb7 at 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 at 192.168.0.104 for seqno 102 (Critical Request)
>Destroying call '5ac70b3e78833fd40669cb867ba3ccb7 at 192.168.0.104'
>
>
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-- 
Scott Henderson
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