[Asterisk-Users] SIP Peers unreachable

[Uriel Carrasquilla]

I have exactly the same problem using Xten.  I have tried with different
codecs such as ulaw, 711 and gsm.  My extension does ring and after two
rings it hangs up.

-----Original Message-----
From: asterisk-users-admin at lists.digium.com
[mailto:asterisk-users-admin at lists.digium.com]On Behalf Of Chris
Sent: Friday, May 02, 2003 12:42 PM
To: asterisk-users at lists.digium.com
Subject: [Asterisk-Users] SIP Peers unreachable


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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