Starting out in voxalot
Hi guys.. I speak from Brazil and have been using Voipdiscount service with my Addpac ap190 voip gateway for outbound calls to PSTN lines...
Recently I installed the same setup I use on both my uncle's and my sister's house, so I was trying to figure out a way to talk for free between these 3 locations... My gateway apparently doesn't support many stuff as fancy dial-plans or different servers (to use sipbroker) so I'm giving a shot with Voxalot, it looks very promising! My problems: I set up my voipdiscount account in the member panel, it says registered. I can dial 600 from my phone, but can't dial any outbound pstn, neither sipbroker (its already configured by default after dialing * I presume ? ) Sipbroker would be directly available just like the 600 and 500 numbers? Or any setup needed? I am also not getting inbound rings after dialing my sip-logged user in voipdiscount software. BUT if I login in a softphone with my voxalot account, then try dialing from voipdiscount software, it goes through ok! So inbound does seem like a problem in my ATA config Thanks in advance, any help appreciated! |
For calls via sipbroker, you use an *, but a call to 500 and 600 or any other voxalot number should be dialed without the asterisk.
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When I dial *011552 (alias i made for my sister acc logged in softphone) I get fast busy signal in my phone
Same with *393613 (echo test I think) I'm not at home now, later I'll post the SIP error codes caught by ethereal if those are of any help Thank you |
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Priority Pattern Replacement Provider Active |
i got this from my ATA debug feature when trying to make a PSTN call
I am registered to voxalot with my ATA, and am registered to voipdiscount in voxalot "providers" section Anyone know what this may be? An issue with voipdiscount? I can dial a voxalotuser directly through his 6 digit username I cannot dial the same voxalot user through a sipbroker alias (*011554 for instance) I cannot dial any PSTN calls through voipdiscount, 407 proxy authentication required, but I thought voxalot was doing it for me through their server? Isn't that the purpose? All I need is: outgoing calls through voipdiscount PSTN, incoming calls using alias for hardware typing (sipbroker or voxalot) 798 <Call 18> : Initiate callee with dial-peer([0-9]T) status(CalleeDe terminedAll) id(b1800-56cf-40a-802f-005e1bc98) 799 <NetEP 18> : InitiateOutCall: calledNum(00553191153937) callingNum( ) target(sip-server) 800 <NetEP 18> : DoCall: calledAddr(sip:00553191153937@voxalot.com:5060 ) callingAddr() 801 <SIP 0> : No authentication information available 802 <SIP 18> : Send INVITE Request 803 <SIP 18> : Transaction Client (24 INVITE) Timeout (retry #1) 804 <SIP 18> : Send INVITE Request 805 <SIP 18> : Receive 407 Proxy Authentication Required 806 <SIP 18> : Transaction (24 INVITE) completed 807 <SIP 18> : Send ACK Request 808 <SIP 0> : No opaque in authentication 809 <SIP 0> : Adding authentication information 810 <SIP 18> : Send INVITE Request 811 <SIP 18> : Receive 407 Proxy Authentication Required 812 <SIP 18> : Receive 407 Proxy Authentication Required Response again 813 <SIP 18> : Transaction Client (25 INVITE) Timeout (retry #1) 814 <SIP 18> : Send INVITE Request 815 <SIP 18> : Receive 403 Use From=id next time 816 <SIP 18> : Transaction (25 INVITE) completed 817 <SIP 0> : Adding authentication information 818 <SIP 18> : Send ACK Request 819 <SIP 18> : Check Event Relation 820 <SIP 18> : ReleaseWithNothing 821 <Call 18> : Terminated from(fffffffe) this(Remote:Unknown) before(NULL) forced(0) 822 <CEP 000000> : DisconnectCall at Busy 823 <CEP 000000> : StopSignal 824 <CEP 000000> : Disconnect (0) 825 <NetEP 18> : Call TO <sip:00553191153937@voxalot.com> terminated reason(Remote:Unknown) 826 <SIP 18> : Receive 403 Use From=id next time 827 <SIP 18> : Receive 403 Use From=id next time Response again 828 <SIP 17> : Set Terminated Success for 22 INVITE 829 <SIP 17> : Set Terminated Success for 23 INVITE |
I was typing my last message when you posted Jorge, I'll be looking into your suggestions and make a new post if it works, thanks!
edit: I tried adding the dial-plan _00. ${EXTEN} voipdiscount, still cant make outgoing calls, getting the same error as above... when registering to voipdiscount directly in ata, the call goes through normally. Is there any port possibly blocking anything? I am behind a speedstream router, forwarding port 5060 UDP to adapter, I've tried DMZ mode also but no help. I've read accross the forums that betamax providers are supposed to work (unless web-callback that needs 2 legs connected), I don't know what could be wrong... I also never can make my ATA ring, but thats a config problem with it tho... I'm emailing their support team. |
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Hmm where should I do this? What does this number do?
What is configured in my ata are those: sip-server voxalot.com sip-username <voxalot login> sip-password <pass> Edit: I looked around the config console and found a setting called CLID that wasn't set, I tried putting my login number there and its now connecting PSTN calls! You just nailed it Kars :) Thanks for all the help guys Is anyone familiar with addpac ap190 ? this is its pdf sheet http://www.addpac.com/pdf/APOS_eng.pdf I cant seem to make inbound calls work in any way, they just go to my voxalot voice mail |
I was trying to ring my ATA and logged the debug report, here it is, what could it be? Is there any specific setting I should be looking to make inbound calls possible?
2 <SIP 23> : Receive INVITE Request 3 <NetCon 23> : Using inbound voice peer(voip 100) by answer-address match 4 <Call 23> : From Net - calledParty() callingParty(tanianobre) 5 <Call 23> : Terminated from(fffffff7) this(Local:InvalidNumber) before(NULL) forced(0) 6 <NetEP 23> : Call TO <tanianobre> terminated reason(Local:InvalidNumber) 7 <SIP 23> : Transaction Server (5 INVITE) Timeout (retry #1) 8 <SIP 23> : Send 404 Response 9 <SIP 23> : Transaction Server (5 INVITE) Timeout (retry #2) 10 <SIP 23> : Send 404 Response 11 <SIP 23> : Transaction Server (5 INVITE) Timeout (retry #3) 12 <SIP 23> : Send 404 Response 13 <SIP 23> : Transaction Server (5 INVITE) Timeout (retry #4) 14 <SIP 23> : Send 404 Response 15 <SIP 23> : Transaction Server (5 INVITE) Timeout (retry #5) 16 <SIP 23> : Send 404 Response 17 <SIP 23> : Transaction Server (5 INVITE) Timeout (retry #6) 18 <SIP 23> : Send 404 Response |
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