Thanks for the response Joseph. It ended up that Telasip needed to make a change on there end. They needed to disable re-invites. BTW, I wanted to give a big plug for Telasip. I thought when I called they would simply tell me it was my problem and they did not support asterisk. This was not the case at all. I recieved promt friendly curtious service. These guys had my problem fixed within 15 min of sending them my log file. I cannot say enought good things about them right now.
________________________________ From: [EMAIL PROTECTED] on behalf of Joseph Tanner Sent: Thu 3/9/2006 11:22 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk @ Home 2.6 Call hangs up My guess, is nat problems. Just for fun, try dialing your inbound number from something not connected to that asterisk box, say a cellphone. I know you're using IAX and SIP, so you'd think you wouldn't run into a double-nat problem (nat going out, nat coming in), but you never know. I have odd issues pop up sometimes when I try calling from my asterisk box right back into it, and I don't even have any nat in the way. Do outgoing calls generally work fine? How do incoming calls work when dialing from an outside line? For the heck of it, try calling out normally, and use a cellphone (or whatever) to dial into the asterisk box. Can it handle an outgoing AND incoming call at the same time, as long as it's not calling itself? If incoming calls still fail, then look into nat issues. Perhaps you can permanently forward port 5060 or 5061 (whichever you use, probably 5060) to your asterisk box, see if that helps any. May need to forward ports 1000-2000 as well. Joseph Tanner On 3/9/06, Jerry Rasmussen <[EMAIL PROTECTED]> wrote: > > > I have installed asterisk @ home 2.6. I am using a Telasip VOIP account. > When I make outbound or inbound calls the calls seem to connect and then get > hung up. I was wondering if there was something that I am misisng. I have > tried several different sip.conf configurations. Here is what they are > currently. > > > telasip-gw > context=telasip-in > dtmfmode=rfc2833 > fromuser=jrasxxx > host=gw4.telasip.com > insecure=very > nat=yes > secret=xyz > type=peer > username=jrasxxx > > 5555551212 > context=from-pstn > dtmfmode=rfc2833 > host=gw4.telasip.com > insecure=very > nat=yes > qualify=yes > secret=xyz > type=peer > username=jrasxxx > > The odd thing is it worked once or twice then stopped. If anyone could shed > some light it would be greatly apperciated. > > Here is what the asterisk output looks like: > -- AGI Script fixlocalprefix completed, returning 0 > -- Executing SetVar("IAX2/100-2", "OUTNUM=7705555555") in new stack > -- Executing Cut("IAX2/100-2", "custom=OUT_2|:|1") in new stack > -- Executing GotoIf("IAX2/100-2", "0?16") in new stack > -- Executing Dial("IAX2/100-2", "SIP/telasip-gw/7705555555") in new > stack > -- Called telasip-gw/7705555555 > -- SIP/telasip-gw-3091 is ringing > -- SIP/telasip-gw-3091 answered IAX2/100-2 > == Spawn extension (macro-dialout-trunk, s, 14) exited non-zero on > 'IAX2/100-2' in macro 'dialout-trunk' > == Spawn extension (from-internal, 7705555555, 1) exited non-zero on > 'IAX2/100-2' > -- Executing Macro("IAX2/100-2", "hangupcall") in new stack > -- Executing ResetCDR("IAX2/100-2", "w") in new stack > -- Executing NoCDR("IAX2/100-2", "") in new stack > -- Executing Wait("IAX2/100-2", "5") in new stack > -- Executing Hangup("IAX2/100-2", "") in new stack > == Spawn extension (macro-hangupcall, s, 4) exited non-zero on > 'IAX2/100-2' in macro 'hangupcall' > == Spawn extension (from-internal, h, 1) exited non-zero on 'IAX2/100-2' > -- Hungup 'IAX2/100-2' > > > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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