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PlusNet voip and Asterisk

timsmith
Grafter
Posts: 39
Registered: 15-01-2009

PlusNet voip and Asterisk

Hi,
Im hoping someone can help as this is making me tear my hair out now!
Ive got an asterisk box installed at home with a routable IP address. The two sipgate VOIP trunks I have configured work perfectly. I recently took up PlusNet Broadband Phone and cannot get the trunk to work reliably.
Without changing the configuration, sometimes calls go through perfectly (to the "weasels test app") and other times, I get messages on the asterisk console that there is no extension configured.  As my asterisk box is continually on and nothing changes between these test calls, I can only assume that something is different in the SIP packetst that PlusNet are sending.
Im pretty new to Asterisk and SIP so Im struggle to suss this one out and would appreciate any help.
the relevant excerpts from sip.conf and extensions.conf are below. At the moment teh call is supposed to go to a Conference Bridge with meetme. As with weasels, sometimes this works and other times it doesnt.
[general]                                                                                                                                         
                                                                                                                                                 
register=778XXXX:XXXXXX@sip2.plus.net/778XXXX                                                                                                   
defaultexpiry=60                                                                                                                                 
                                                                                                                                                 
[778XXXX]                                                                                                                                         
type=friend                                                                                                                                       
host=sip2.plus.net                                                                                                                               
canreinvite=no                                                                                                                                   
nat=no                                                                                                                                           
secret=XXXXXX                                                                                                                                 
username=778XXXX                                                                                                                                 
fromuser=778XXXX                                                                                                                               
qualify=30000                                                                                                                                     
dtmfmode=inband                                                                                                                                   
allow=ulaw                                                                                                                                       
allow=gsm                                                                                                                                         
allow=alaw                                                                                                                                       
allow=g726                                                                                                                                       
context=incoming_778XXXX                                                                                                                         
fromdomain=sip2.plus.net                                                                                                                         
insecure=port,invite
                           
[incoming_778XXXX]
exten => s,1,NoOp()
exten => s,n,Verbose(1|Extension XXXX)
exten => s,n,MeetMe(6000,i,54321)
exten => _778XXXX,1,NoOp()
exten => _778XXXX,n,Verbose(1|Extension XXXX)
exten => _778XXXX,n,MeetMe(6000,i,54321)
Many thanks
Tim
3 REPLIES
Community Veteran
Posts: 6,262
Thanks: 433
Fixes: 39
Registered: 30-07-2007

Re: PlusNet voip and Asterisk

Not had any experience of Asterix but AFAIK  your domain should really be sip.plus.net ( not sip2 ). sip2 is going to be retired soon although I dont think it is yet..
See if that makes any difference.
timsmith
Grafter
Posts: 39
Registered: 15-01-2009

Re: PlusNet voip and Asterisk

Thanks for the reply.
sip.plus.net does seem to work better in that all calls do actually get presented properly to Asterisk. However, some calls go to the extension matching my username in the defualt context and others go to the extension matching my username in the correct context specified in the peer configuration.
Now, I can just direct these to the same place, but there must be differences in what plusnet are sending that is causing this.
Can anyone advise?
Thanks
Tim
Community Veteran
Posts: 6,262
Thanks: 433
Fixes: 39
Registered: 30-07-2007

Re: PlusNet voip and Asterisk

Quote
However, some calls go to the extension matching my username in the defualt context and others go to the extension matching my username in the correct context specified in the peer configuration.

I'm afraid you've lost me there , I'm familiar with a fair amount of voip kit but unfortunately not Asterix.
However have you checked out this post http://community.plus.net/forum/index.php/topic,60696.0.html and this link http://community.plus.net/forum/index.php/topic,60173.msg491058.html#msg491058 from it.
That last link seems to suggest a different set of allowed codecs from the ones you have , that might be worth a try.