asterisk users - Apr 2005 - problems with call-forward from ccme to * on sip trunk

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hi folks

I've a strange problem, probably a mistake but I don't see it :(

Description:

My ephone-dn number on ccme, that is a simple connection plar for all 
ISDN calls, is 601
The voicemailmain on asterisk is 5900.
CCME: 192.168.17.1
*: 192.168.17.10

My sip.conf: http://www.pastebin.com/266718
My extension.conf: http://www.pastebin.com/266720
My voicemail.conf: http://www.pastebin.com/266722

when I call the asterisk server from SIP free accounts, I receive the 
call on 601 (my 7960 phone) and then the call will be forwarded to 
voicemail without any problem.
But when I receive a call from ISDN cloud, the 601 rings, the call is 
forwarded (see debug) on voicemail (number 5601), but the line goes 
down.

This is the debug, that is I suppose the problem is on my Asterisk 
config (the 'ext-number' is the caller ID): 
http://www.pastebin.com/266724

I hope you could help me :)
Thanks for all
Regards
Andrea
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v1.2.4 (Darwin)

iD8DBQFCUIYAMakHrsrHP9wRAlR1AKDKNzARotrmFMPphvjwqjp8da4SwACfQ6lo
hxesZUu9t220j8zfQHW2DX0=zJCw
-----END PGP SIGNATURE-----

I think this problem is exactly the one I am having.

The issue is:
http://www.pastebin.com/266724
042 Found no matching peer or user for '192.168.17.1:56730'
to which asterisk generates a "SIP/2.0 404 Not Found" (line 057)

yet you have it configured here:

[operator]
type=peer
canreinvite=no
host=192.168.17.1
context=cme-pbx

Hopefully someone with a working configuration can provide feedack.

Regards,

Nathan.


Andrea Riela wrote:
> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
> 
> Hi folks
> 
> I've a strange problem, probably a mistake but I don't see it :(
> 
> Description:
> 
> My ephone-dn number on ccme, that is a simple connection plar for all 
> ISDN calls, is 601
> The voicemailmain on asterisk is 5900.
> CCME: 192.168.17.1
> *: 192.168.17.10
> 
> My sip.conf: http://www.pastebin.com/266718
> My extension.conf: http://www.pastebin.com/266720
> My voicemail.conf: http://www.pastebin.com/266722
> 
> when I call the asterisk server from SIP free accounts, I receive the 
> call on 601 (my 7960 phone) and then the call will be forwarded to 
> voicemail without any problem.
> But when I receive a call from ISDN cloud, the 601 rings, the call is 
> forwarded (see debug) on voicemail (number 5601), but the line goes down.
> 
> This is the debug, that is I suppose the problem is on my Asterisk 
> config (the 'ext-number' is the caller ID):
http://www.pastebin.com/266724
> 
> I hope you could help me :)
> Thanks for all
> Regards
> Andrea
> -----BEGIN PGP SIGNATURE-----
> Version: GnuPG v1.2.4 (Darwin)
> 
> iD8DBQFCUIYAMakHrsrHP9wRAlR1AKDKNzARotrmFMPphvjwqjp8da4SwACfQ6lo
> hxesZUu9t220j8zfQHW2DX0> =zJCw
> -----END PGP SIGNATURE-----
> 
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Uhmm ...

maybe a connection plar from ccme to an * number (like 511 on my conf), 
then a simple forward from 511 to 601 on ccme?
Something like:

exten => _511,1,Dial(SIP/601,45)

I need help ... :D
Andrea
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v1.2.4 (Darwin)

iD8DBQFCUPWwMakHrsrHP9wRAuSVAKCMyKYIVSP8B+Tc0losELtmJovsEQCcDoOi
gp1ZxZqe+G9hdAK6nEoqlaI=D68e
-----END PGP SIGNATURE-----

This may help:

http://voip-info.org/tiki-index.php?page=Asterisk+Cisco+CallManager+Express+Integration

Is there any debug form your cisco router ? (debug voip event-log or 
similar) or the Asterisk console ?


Nathan.


Andrea Riela wrote:
> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
> 
> Hi folks
> 
> I've a strange problem, probably a mistake but I don't see it :(
> 
> Description:
> 
> My ephone-dn number on ccme, that is a simple connection plar for all 
> ISDN calls, is 601
> The voicemailmain on asterisk is 5900.
> CCME: 192.168.17.1
> *: 192.168.17.10
> 
> My sip.conf: http://www.pastebin.com/266718
> My extension.conf: http://www.pastebin.com/266720
> My voicemail.conf: http://www.pastebin.com/266722
> 
> when I call the asterisk server from SIP free accounts, I receive the 
> call on 601 (my 7960 phone) and then the call will be forwarded to 
> voicemail without any problem.
> But when I receive a call from ISDN cloud, the 601 rings, the call is 
> forwarded (see debug) on voicemail (number 5601), but the line goes down.
> 
> This is the debug, that is I suppose the problem is on my Asterisk 
> config (the 'ext-number' is the caller ID):
http://www.pastebin.com/266724
> 
> I hope you could help me :)
> Thanks for all
> Regards
> Andrea
> -----BEGIN PGP SIGNATURE-----
> Version: GnuPG v1.2.4 (Darwin)
> 
> iD8DBQFCUIYAMakHrsrHP9wRAlR1AKDKNzARotrmFMPphvjwqjp8da4SwACfQ6lo
> hxesZUu9t220j8zfQHW2DX0> =zJCw
> -----END PGP SIGNATURE-----
> 
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users