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Unread 07-17-2006, 05:34 AM   #1
nobre84
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Default 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!
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Unread 07-17-2006, 05:50 AM   #2
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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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Unread 07-17-2006, 03:26 PM   #3
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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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Unread 07-18-2006, 03:47 AM   #4
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Quote:
I set up my voipdiscount account in the member panel, it says registered.
I assume you do not have a voipdiscount voip-in number and you use that provider to make outbound calls only, so you do not need to enable registration. It doesn't hurt, but without an inbound number, registering is actually of no use. The registered status is an indication that you will be able to receive calls thorugh that provider, not an indication that you will be able to make outgoing calls.

Quote:
I can dial 600 from my phone, but can't dial any outbound pstn, neither sipbroker (its already configured by default after
Have you created a dial plan in your Voxalot account? If you have not already, your calls will not be routed to your paid providers which are the only ones which can connect to PSTN and calls will fail. A simple dialplan for Betamax providers like sipdiscount would be:
Code:
Priority	Pattern  	Replacement  	Provider  	Active  	 
1 		_00. 		${EXTEN} 	sipdiscount 	Yes 	[Edit] [Delete]
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Unread 07-18-2006, 04:19 AM   #5
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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
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Unread 07-18-2006, 04:32 AM   #6
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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.

Last edited by nobre84; 07-18-2006 at 05:04 AM.
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Unread 07-18-2006, 02:18 PM   #7
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Quote:
Originally Posted by nobre84
826 <SIP 18> : Receive 403 Use From=id next time
Ah yes, I've seen this. Try to set your voxalot number as "Phone number" in your ATA, not your PSTN-to-SIP provider access number.
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Unread 07-18-2006, 03:14 PM   #8
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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

Last edited by nobre84; 07-18-2006 at 03:39 PM.
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Unread 07-19-2006, 05:35 PM   #9
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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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