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...[GoTalk] username=09xxxxxx fromuser=09xxxxxx fromdomain=sip.gotalk.com type=peer secret=xxxxxxxx qualify=yes host=sip.gotalk.com disallow=all allow=g729 ;GoTalk Inbound [09xxxxxx] username=09xxxxxx type=user secret=xxxxxxxx fromuser=09xxxxxx host=sip.gotalk.com context=from-vsp canredirect=no Registration string is register=09xxxxxx:xxxxxxxx at sip.gotalk.com/09xxxxxx David Klaverstyn Systems Administrator Information Services, Asia-Pacific Intergraph Corporation Level 3, 299 Coronation Drive Milton, QLD 4064 AU P 61.7.3510.8951 F 61.7.3510.8901 david.klaverstyn at intergra...

..._00339xxxxxxxx,7,Set(CALLERPRES()=prohib_not_screened) exten => _00339xxxxxxxx,8,Dial(SIP/MyOperator/${NUMAPPEL},180,rt) exten => _00339xxxxxxxx,9,Hangup and i have in sip.conf: [MyOperator] type=peer host=host-of-my-operator qualify=yes dtmf=rfc2833 nat=no canreinvite=no canredirect=yes insecure=port,invite dtmfmode=rfc2833 disallow=all allow=g729 allow=alaw allow=g723 defaultuser=0033xxxxxx secret=xxxxx When i call directly from [MyOperator], no probleme i have sound/Voice but when a customer call to the "00339xxx..", the call are correct, asterisk call to my sta...

...g calls register => ##########:secret@sip.stanaphone.com/1000 register => ##########:secret@sip.provider.net/4078 register => ##########:secret@sip.provider.net/4077 [stanaphone-out] ;works!!! host=sip.stanaphone.com context=sip type=friend dtmfmode=rfc2833 canredirect=no disallow=all allow=ulaw insecure=very username=secret fromuser=secret secret=secret ;more testing broadvoice examples ;THIS ONE WORKS!!! [our-sip-provider-out] type = peer host = sip.provider.net secret = secret user=phone ; I needed this to make it work (what tha ????) fromuser = secret user...

...tly?"free calls". ? I am using full cone nat at my PIX. Can anyone give me an explanation of what may justify it or a possible solution my sip.conf [vpb] type=peer secret=xxxx username=xxxxx fromuser=xxxxx host=sip1.voipbuster.com fromdomain= sip1.voipbuster.com insecure=very canredirect=no disallow=all allow=gsm allow=ulaw nat=yes qualify=no context=internal externip=my.public.ip.address localnet=my.local.ip.address /my.local.subnet.mask? Many thanks! Rafael

Hi, I've set up FreeWorldDialup on my asterisk server but when I dial the service numbers, I get message '486 Busy Here '. When I dial any other number, it says 'All Circuits are busy now'. What is the problem with my settings. I've followed all the instructions step by step. Zeeshan -------------- next part -------------- An HTML attachment was scrubbed... URL:

...eex allow=ulaw register=myusername:password@calamar0.nikotel.com/calamar0.nikotel.com [calamar0.nikotel.com] secret=password username=myusername fromuser=myusername fromdomain=nikotel.com qualify=yes type=friend context=waehlplan_eingehend_nikotel host=calamar0.nikotel.com nat=yes canreinvite=yes canredirect=no promiscredir=yes insecure=very ;incominglimit=1 ;restrictcid=yes callerid=myusername@calamar0.nikotel.com disallow=all allow=gsm [10] type=friend username=10 secret=xx host=dynamic dtmfmode=rfc2833 nat=no qualify=1000 mailbox=10@10 callgroup=2 pickupgroup=2 disallow=all allow=gsm ############...

Let me explain: When I dial into Asterisk ( I have a SIP trunk - which I need to make sure is not faulty), I only get one-way voice communication. The calling party, from the SIP trunk hears nothing - the extension rings on the Asterisk server (you can see it in the CLI and hear it at the computer), and the softphone rings However, when you answer the SIP softphone , you can only hear the

Sipdiscount has replaced their asterisk servers for another thing. Then, no more iax. Ok, but I can't make calls using sip also... I'm getting a "forbidden" error when using sip1.sipdiscount.com. Anybody got it working? -- Alejandro Vargas

I'm going to try and ask this again and keep it short and as too the point as I can while still providing enough info to be of use. PLEASE advise if I am going about this wrong or asking too much. I'm seriously doing my BEST to throughly read the docs and try a bunch of things BEFORE coming here to ask and possibly annoy. If is documentation that explains thsi process in terms that