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

[Scott Henderson]

I am guessing that the core of your problem here is that the phone isn't 
registering with the server (a statement of the obvious).   Anyway I 
just went through this on a project and if you study the debugs, you 
will see where the registration falls down.  Mine was a series of typos 
on both the phones configuration and the .conf files.  It took look at 
the debugs repeatedly to track down all the little errors that were 
between my and the registration. 

Also take a look as "sip show peers" and "sip show users" to see what 
you can see there.

Scott

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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