Yannick Posted October 3, 2011 Report Share Posted October 3, 2011 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. Quote Link to comment Share on other sites More sharing options...
pbx support Posted October 3, 2011 Report Share Posted October 3, 2011 It could be that they(83.143.188.165:5060) don't like the SIP INFO for AOC. INFO sip:SIP_5Fc@83.143.188.161:5060;bnat=yes SIP/2.0 Via: SIP/2.0/UDP 62.148.184.139:5060;branch=z9hG4bK-adff363c2097d25173b76ddf330abea5;rport Route: <sip:83.143.188.165;lr=on;ftag=1949662168> From: <sip:31707113070@sip1.budgetphone.nl;user=phone>;tag=f7c6a93f36 To: <sip:0653667420@83.143.188.161;user=phone>;tag=1949662168 Call-ID: 1427118654@83.143.188.161 CSeq: 17969 INFO Max-Forwards: 70 Contact: <sip:31707113070@62.148.184.139:5060;transport=udp> Content-Type: application/vnd.etsi.aoc+xml Content-Length: 302 <?xml version="1.0" encoding="UTF-8"?> <aoc><aoc-d><charging-info>subtotal</charging-info><recorded-charges><recorded-currency-units><currency-id>USD</currency-id><currency-amount>0.01</currency-amount></recorded-currency-units></recorded-charges><billing-id>normal-charging</billing-id></aoc-d></aoc> [7] 2011/10/03 19:26:39: SIP Rx udp:83.143.188.165:5060: SIP/2.0 400 Bad Request Via: SIP/2.0/UDP 62.148.184.139:5060;received=62.148.184.174;branch=z9hG4bK-adff363c2097d25173b76ddf330abea5;rport=5060 From: <sip:31707113070@sip1.budgetphone.nl;user=phone>;tag=f7c6a93f36 To: <sip:0653667420@83.143.188.161;user=phone>;tag=1949662168 Call-ID: 1427118654@83.143.188.161 CSeq: 17969 INFO Content-Length: 0 [7] 2011/10/03 19:26:39: Call 1427118654@83.143.188.161: Clear last request [7] 2011/10/03 19:26:44: SIP Rx udp:83.143.188.165:5060: BYE sip:31707113070@62.148.184.174:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 83.143.188.165;branch=z9hG4bK1828.fa929f06.0 Via: SIP/2.0/UDP 83.143.188.161:5060;rport=5060;received=83.143.188.161;branch=z9hG4bK999006433 From: <sip:0653667420@83.143.188.161;user=phone>;tag=1949662168 To: <sip:31707113070@sip1.budgetphone.nl;user=phone>;tag=f7c6a93f36 Call-ID: 1427118654@83.143.188.161 CSeq: 21 BYE Max-Forwards: 12 Reason: Q.850 ;cause=16 ;text="Normal call clearing" Content-Length: 0 Quote Link to comment Share on other sites More sharing options...
Yannick Posted October 4, 2011 Author Report Share Posted October 4, 2011 Thanks! Should I ask 83.143.188.165 (Budgetphone) to support it or is there a way to disable it in Snom? Also I tried the same thing from a different incoming line (voxalot) but got an "unsupported media type" error: http://pastebin.ca/2086644 How could I find out which media type is unsupported and how can I exclude it? Quote Link to comment Share on other sites More sharing options...
pbx support Posted October 4, 2011 Report Share Posted October 4, 2011 Looks like no one supporting advice of charge INFO messages. I am not sure if we have a way to disable sending this AOC INFO message. Quote Link to comment Share on other sites More sharing options...
Yannick Posted October 5, 2011 Author Report Share Posted October 5, 2011 Would it help if I would forward my incoming sip phone numbers to an intermediate service or something? I'm quite at loss as how to proceed now. Any other software you would recommend? Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted October 5, 2011 Report Share Posted October 5, 2011 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. Quote Link to comment Share on other sites More sharing options...
Yannick Posted October 6, 2011 Author Report Share Posted October 6, 2011 Thanks, how do I remove the trunk rates? I had to put them in place as before I got the message: "extension doesn't have enough credit to place call" Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted October 6, 2011 Report Share Posted October 6, 2011 Eh yea seems you went down to road of billing... Do you need that? I would remove everything that has to do with billing: rates, credit in the domain and the extension. Quote Link to comment Share on other sites More sharing options...
pbx support Posted October 7, 2011 Report Share Posted October 7, 2011 "extension doesn't have enough credit to place call" Did you clear text box in the "Trunk->Rates" field and then try? Quote Link to comment Share on other sites More sharing options...
Yannick Posted October 8, 2011 Author Report Share Posted October 8, 2011 thanks, I removed the rate info and now that specific error went away. Unfortunately I still get the "we're sorry your call could not be connected" message. This is the log I get now: http://pastie.org/2659758. Looks ok to me... Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted October 8, 2011 Report Share Posted October 8, 2011 This is the log I get now: http://pastie.org/2659758. Looks ok to me... The link does not work for me... Can you attach it here? Quote Link to comment Share on other sites More sharing options...
pbx support Posted October 8, 2011 Report Share Posted October 8, 2011 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. Quote Link to comment Share on other sites More sharing options...
Yannick Posted October 9, 2011 Author Report Share Posted October 9, 2011 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 Quote Link to comment Share on other sites More sharing options...
Recommended Posts
Join the conversation
You can post now and register later. If you have an account, sign in now to post with your account.
Note: Your post will require moderator approval before it will be visible.