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outgoing calls on betamax/voxalot


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I'm new to this and trying to setup an IVR that will redirect calls to mobile lines on menu selection.

I'v got the IVR working but when it should redirect to my mobile line it's giving me the default betamax error message: "sorry your call could not be connected.". I'v setup credits for the domain and a dial plan that selects the correct betamax trunk but then when redirecting it gives me this message.

I don't know why this happens but it seems that either the forwarded number is incorrect (missing country code or something or appending headers to the address) or the redirected caller id is not accepted. I've tried many number formats (with leading 00's, +, etc) without avail.

I don't know how to debug this and could use some help.


Here is my SIP log: http://pastebin.ca/2086488


Could it be that it is trying to find a user at eu.voxalot.com instead of calling the mobile number?


Any help on how to debug this is very welcome.

I tried doing more debugging by trying to call out using an iPhone sip client but then the other party could hear me but I couldn't hear anything.

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It could be that they( don't like the SIP INFO for AOC.


INFO sip:SIP_5Fc@;bnat=yes SIP/2.0
Via: SIP/2.0/UDP;branch=z9hG4bK-adff363c2097d25173b76ddf330abea5;rport
Route: <sip:;lr=on;ftag=1949662168>
From: <sip:31707113070@sip1.budgetphone.nl;user=phone>;tag=f7c6a93f36
To: <sip:0653667420@;user=phone>;tag=1949662168
Call-ID: 1427118654@
CSeq: 17969 INFO
Max-Forwards: 70
Contact: <sip:31707113070@;transport=udp>
Content-Type: application/vnd.etsi.aoc+xml
Content-Length: 302

<?xml version="1.0" encoding="UTF-8"?>
[7] 2011/10/03 19:26:39:        SIP Rx udp:
SIP/2.0 400 Bad Request
Via: SIP/2.0/UDP;received=;branch=z9hG4bK-adff363c2097d25173b76ddf330abea5;rport=5060
From: <sip:31707113070@sip1.budgetphone.nl;user=phone>;tag=f7c6a93f36
To: <sip:0653667420@;user=phone>;tag=1949662168
Call-ID: 1427118654@
CSeq: 17969 INFO
Content-Length: 0

[7] 2011/10/03 19:26:39:        Call 1427118654@ Clear last request
[7] 2011/10/03 19:26:44:        SIP Rx udp:
BYE sip:31707113070@;transport=udp SIP/2.0
Via: SIP/2.0/UDP;branch=z9hG4bK1828.fa929f06.0
Via: SIP/2.0/UDP;rport=5060;received=;branch=z9hG4bK999006433
From: <sip:0653667420@;user=phone>;tag=1949662168
To: <sip:31707113070@sip1.budgetphone.nl;user=phone>;tag=f7c6a93f36
Call-ID: 1427118654@
CSeq: 21 BYE
Max-Forwards: 12
Reason: Q.850 ;cause=16 ;text="Normal call clearing"
Content-Length: 0

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If the other side does not support the AOC information, it should (silently) drop it. Actually, technically, 400 on a INFO does not terminate the dialog and it does not have to.


Anyway, the AOC information probably comes from the trunk rates. If your service provider drops the call on this, it is probably not possible to do rates on this trunk. Try to remove the rates and see if the calls go through.

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The link does not work for me... Can you attach it here?


Just had to remove the dot at the end of the link.


Looking at the trace there are no AOC messages now. Now it is a matter of choosing the suitable "Trunk->Remote Party/Privacy Indication" option. Some providers do not like the P-Asserted header. Try using "Not indication" or "Remote-Party-ID" from the drop-down and place the call again.

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Try using "Not indication" or "Remote-Party-ID" from the drop-down and place the call again.

Wow, that did the trick!

So happy this forum and support (even for free users!) exists. I remember my early days of programming, when you could spend days figuring out 1 simple thing. Now with the help I got a jump start into this great product! thank you

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