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Summary:ASTERISK-06415: Uniden UIP 200 with Asterisk V 1.2.X
Reporter:Michael G (mgichoga)Labels:
Date Opened:2006-02-24 13:55:48.000-0600Date Closed:2011-06-07 14:10:06
Priority:MinorRegression?No
Status:Closed/CompleteComponents:Channels/chan_sip/General
Versions:Frequency of
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Description:I have discovered that when running asterisk version 1.2.X with the Uniden UIP200 that they will not ring until in their extension portion in the sip.conf qualify is set to e.g

[200]
type=friend
secret=mysecrte
host=dynamic
context=default
dtmfmode=rfc2833
callerid=Line1
nat=never
qualify=no
callgroup=1
pickupgroup=1

After qualify is set to no it rings. The reason I'm submitting this is becuase I've traced a problem with intercom system which uses em signaling and these Uniden phones will disconnect after 45 seconds because asterisk is not monitoring. They seem to be working fine with regular calls. Version 1.0.9 doesn't seem to be having these issues.
Comments:By: Russell Bryant (russell) 2006-02-26 18:48:15.000-0600

Can you provide 'sip debug' output that shows that with qualify=yes, the phone is actually responding to our OPTIONS message?  If the phone is not responding to OPTIONS, then this would be expected behavior because this indiates to Asterisk that the phone is unreachable.
By: Michael G (mgichoga) 2006-02-27 14:47:11.000-0600

This is what I got from the sip debug <peer>

Retransmitting #4 (NAT) to 172.16.18.4:5060:
OPTIONS sip:702@172.16.18.4:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.0.6:5060;branch=z9hG4bK5c0766d9;rport
From: "asterisk" <sip:asterisk@172.16.0.6>;tag=as133a204c
To: <sip:702@172.16.18.4:5060>
Contact: <sip:asterisk@172.16.0.6>
Call-ID: 595ecb2966c7e8887686470a66b22014@172.16.0.6
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 27 Feb 2006 21:31:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
By: Michael G (mgichoga) 2006-02-27 14:49:41.000-0600

also forgot to mention this, when the qualify is set to 1000 this warning message constantly appears

app_dial.c:1011 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination)
By: Russell Bryant (russell) 2006-02-27 14:59:30.000-0600

> Retransmitting #4 (NAT) to 172.16.18.4:5060:

If the OPTIONS message is having to be retrasmitted like that, there are only two possibilities.  Either the phone is truly unreachable, or the phone does not support OPTIONS.  The latter appears to be the case here.  In that case, the only option you have to make this phone work is to turn off qualify.  

The behavior you are seeing is expected.  I would encourage you to request that the phone vendor implement this feature, as it is extremely easy to do.