[Asterisk-Users] SIP Peers unreachable

[Uriel Carrasquilla]

I am the second guy with the same problem.
The Xten SIP phone registers OK as soon as I launch it from W2K (Windows).
Then I dial a Zap extension off a channel bank (that is working very well
with calls from PSTN and from other Zap locally or remotely via IAX).  The
Phone rings twice and the Spawn the disconnect.  If I pick up, the it hangs
up anyway.
Uriel

-----Original Message-----
From: asterisk-users-admin at lists.digium.com
[mailto:asterisk-users-admin at lists.digium.com]On Behalf Of John Todd
Sent: Friday, May 02, 2003 1:33 PM
To: asterisk-users at lists.digium.com
Subject: Re: [Asterisk-Users] SIP Peers unreachable



I would suggest firing up "tethereal port 5060" on the command line
in a different window and see if your SIP requests are making it to
the destination device when you create a call.  If you don't have
tethereal, get the "ethereal" packages and install them.

Remove the "qualify" lines until you get things working.

Launch asterisk with "asterisk -vvvgc" to look at some additional
debugging information when attempting to dial.

What happens when you dial 444 from one of your phones?

JT




>Hi Everyone,
>
>I'm new to * and I'm trying to setup a small configuration of SIP clients.
>Eventually when I get this working I plan on expanding with a Digium
>developers kit to add analog phones and PSTN access.
>
>My two end points are an Xten softphone and a Mitel 5055 SIP phone. Both
>peers seem to register with * but I cannot call to one another. When I dial
>the associated extension, the call goes to the programmed voicemail
>extension (busy) yet if I create an extension to call out through the proxy
>(IX66), I can still reach my destination. It's just calling within * there
>is a problem. I suspect it's because the status is unreachable but I'm not
>sure how to fix it.
>
>Here is the sip show peers output.
>
>Name/username    Host                 Mask             Port     Status
>sipset/sipset    192.200.14.31    (D)  255.255.255.255  5060
UNREACHABLE
>sippc/sippc      192.200.14.33    (D)  255.255.255.255  5060
UNREACHABLE
>
>Here is the sip.conf settings:
>[general]
>port = 5060
>bindaddr = 0.0.0.0
>context = default
>register => 9055551212 at somewhere.homeip.net
>
>[sippc]
>type=friend
>username=sippc
>secret=blah
>host=dynamic
>qualify=3000
>
>[sipset]
>type=friend
>username=sipset
>secret=blah
>host=dynamic
>qualify=3000
>
>Here is the extensions.conf settings:
>exten => 421,1,Dial(SIP/sipset) 	  ; Mitel 5055 SIP Phone
>exten => 421,2,Voicemail(u421)
>exten => 421,102,Voicemail(b421)
>exten => 422,1,Dial(SIP/sippc) 	  ; Xten client
>exten => 422,2,Voicemail(u422)
>exten => 422,102,Voicemail(b422)
>
>exten => 444,1,Dial(SIP/tony at somewhere.homeip.net)  ; friends MSN (4.6)
>account registered to IX66
>
>
>These are the console messages when I dial 421 from 422
>
>     -- Executing Dial("SIP/sippc-b5f6", "SIP/sipset") in new stack
>   == Everyone is busy at this time
>     -- Executing VoiceMail("SIP/sippc-b5f6", "b421") in new stack
>   == Parsing '/etc/asterisk/voicemail.conf':   == Parsing
>'/etc/asterisk/voicemail.conf': Found
>     -- Playing 'vm-theperson'
>     -- Playing 'digits/4'
>     -- Playing 'digits/2'
>     -- Playing 'digits/1'
>     -- Playing 'vm-isonphone'
>     -- Playing 'vm-intro'
>   == Spawn extension (default, 421, 102) exited non-zero on
'SIP/sippc-b5f6'
>
>
>Any help is appreciated.
>
>Thanks.
>
>Chris
>
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