[Asterisk-Users] SIP Peers unreachable

[Mark Spencer]

You might try a "sip debug" to provide something useful.

Mark

On Sat, 3 May 2003, Uriel Carrasquilla wrote:

> 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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