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IPO500 v2 (8.0) issues to receive calls from an ASTERISK via sip trunks

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Jul 29, 2013
45
BR
Hello,

I have an problem to receive calls on IPO500 from a sip trunk with ASTERISK,

I have 8 trunks, 7047 to 7054, but only the 7047 receive calls, all anothers trunk can´t receive calls,

I see the ASTERISK sending the call only to the first trunk (7047) and the PBX send "404 not found".
___________________________________________________________________________________________________________________________
19/10/15 13:12:56-668ms Line = 11, Channel = 1, SIP Message = Response, Direction = From Switch, From = 7088@200.155.13.103, To = 7048@172.20.20.100, Response = 100 Trying
19/10/15 13:12:56-670ms Line = 11, Channel = 1, SIP Message = Response, Direction = From Switch, From = 7088@200.155.13.103, To = 7048@172.20.20.100, Response = 404 Not Found
19/10/15 13:12:56-682ms Line = 11, Channel = 1, SIP Message = Ack, Direction = To Switch, From = 7088@200.155.13.103, To = 7048@172.20.20.100
19/10/15 13:21:10-212ms Line = 11, Channel = 1, SIP Message = Response, Direction = From Switch, From = 7054@200.155.13.103, To = 7050@172.20.20.100, Response = 100 Trying
19/10/15 13:21:10-214ms Line = 11, Channel = 1, SIP Message = Response, Direction = From Switch, From = 7054@200.155.13.103, To = 7050@172.20.20.100, Response = 404 Not Found
19/10/15 13:21:10-228ms Line = 11, Channel = 1, SIP Message = Ack, Direction = To Switch, From = 7054@200.155.13.103, To = 7050@172.20.20.100
____________________________________________________________________________________________________________________________

I think is because the port 5060, but I can´t figure out how to fix this,

Regards,

Emerson.
 
404 not found is the IPO saying that it does not have a route, whether in the SIP URIS (7047, ....up or * for wildcard accept all), or in the DDI table to route the call to the group/extension. the fact that you get this far means all is ok on the sip signalling side, so nothing to change with 5060 and all that.
ps - for sip trace , tick all sip boxes except the HEX outputs
 
Hi dear all,

I change the URI contact and works well,

but the ASTERISKS programmer have to set the extension to a trunk, to receive more than one call at once,

anyone there know how to program an ASTERISK TRUNK?

regards,

Emerson Ribeiro.
 
he he, i knew i recognized it - how's it going Andy? It's been a while since I've been here:)
 
@suportejust - that's incorrect information re the trunk setting and basic call waiting. Sounds like the asterisk programmer isn't that good. Trunk settings, however, aka peering, non-registration, ip authenticated, simply don't use the Registration method. So just untick "Registration Required".
 
Hello,

thanks for the support, the ASTERISK programmer solve the problem,

he program a trunk x trunk and works well,

regards,

Emerson Ribeiro.
 
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