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84 posts

Master Geek


# 80846 5-Apr-2011 06:24
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Hi everyone,

I've been having some issues with Asterisk/FreePBX and trying to get a 2talk SIP trunk to work, until last night I had some major breakthroughs, but one remaining hurdle seems to to be getting in the way ... Outgoing calls.

For testing I have been using X-Lite which seems to be doing ... hmmm ... Okay for the job, but will have to do until I get a proper VoIP phone in the house.

When I attempt to make an outgoing call to my mobile, or any landline for that matter I can see the call heading out in the Asterisk CLI, but about 15 seconds later, I get "All circuits are busy at present, please try again later".

Inbound calls are working perfectly.

For my PEER Details in Asterisk I have ...

regseconds=180
registertimeout=20
register=03xxxxxxx:password@2talk.co.nz/03xxxxxx
port=5060
language=en
dtmfmode=rfc2833
disallow=all
bindport=5060
allowoverlap=no
allow=ulaw&alaw&gsm&ilbc
nat=yes
insecure=very

Does anyone have any idea what may be restricting the calls, and what I should be checking for?

Many thanks.

Alex


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28559 posts

Uber Geek

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  # 455393 5-Apr-2011 08:35
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Do 2talk support gsm and ilbc?


104 posts

Master Geek


  # 455482 5-Apr-2011 12:33
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Try this trunk config:

[Peer Deatils]
type=friend
username=09XXXXXXX
fromuser=09XXXXXXX
secret=password
host=2talk.co.nz
context=from-trunk
dtmfmode=auto
disallow=all
allow=alaw&ulaw
nat=yes
canreinvite=no
insecure=invite,port


Register string:
09XXXXXXX:password@2talk.co.nz/09XXXXXXX



In your original trunk setup, you had the register value under Peer details by the looks.

Take it out of there and put it in the designated field in FreePBX 

That's a tried and true trunk setup - let me know if it doesn't work

Cheers 

 
 
 
 


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  # 455515 5-Apr-2011 13:18
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dtmfmode=auto has the potential to create lots of issues. dtmfmode=rfc2833 should be used unless you know exactly what you're doing.



84 posts

Master Geek


# 455623 5-Apr-2011 18:45
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IT WORKS!

Many thanks for the PEER details config. It works perfectly, incoming and outgoing calls all working perfectly!

Many many many thanks for that!

If I were up there I'd definitely shout a beer or few for that advice. :)

104 posts

Master Geek


  # 455625 5-Apr-2011 18:55
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Great, glad you got it all working.

Steve thanks for the note about dtmf mode.  What sort of issues have you seen with that value?

I have in the past used dtmfmode=rfc2833 but used the auto setting recently as was having some dtmf issues. 

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  # 455638 5-Apr-2011 20:07
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philelvey: Great, glad you got it all working.

Steve thanks for the note about dtmf mode.  What sort of issues have you seen with that value?

I have in the past used dtmfmode=rfc2833 but used the auto setting recently as was having some dtmf issues. 


You should really only ever be using RFC2833 for DTMF and I've never come across any system where RFC2833 didn't work unless things were configured incorrectly somewhere such as incorrect configuration on SIP peers.

Inband obviously can't be used with any codecs apart from G711, the auto mode simply looks for RFC2833 support in the SDP and if that's not there it'll use inband. This can create big problems if you're supporting codecs other than G711 where the scenario of a G.729 connection means DTMF won't work.


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