Yannick

[7] 2011/10/13 22:11:31: SIP Rx udp:83.143.188.165:5060: OPTIONS sip:62.148.184.174:5060 SIP/2.0 Via: SIP/2.0/UDP 83.143.188.165:5060;branch=0 From: sip:pinger@sip1.budgetphone.nl;tag=27c969fc To: sip:62.148.184.174:5060 Call-ID: 8e8757a3-0b35d35c-a61179@83.143.188.165 CSeq: 1 OPTIONS Content-Length: 0 [7] 2011/10/13 22:11:31: SIP Tx udp:83.143.188.165:5060: SIP/2.0 200 Ok Via: SIP/2.0/UDP 83.143.188.165:5060;branch=0 From: <sip:pinger@sip1.budgetphone.nl>;tag=27c969fc To: <sip:62.148.184.174:5060>;tag=4b1ca8abcd Call-ID: 8e8757a3-0b35d35c-a61179@83.143.188.165 CSeq: 1 OPTIONS Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Content-Length: 0 [7] 2011/10/13 22:11:56: Hunt Group 72: Moving to next stage [7] 2011/10/13 22:11:56: Hunt group 72 started 0 calls [7] 2011/10/13 22:11:56: Hunt Group 72: Moving to next stage [7] 2011/10/13 22:11:56: Hunt group 72 started 0 calls [7] 2011/10/13 22:11:56: Hunt Group 72: Moving to next stage [7] 2011/10/13 22:11:56: Hunt group 72 started 0 calls [7] 2011/10/13 22:11:56: Hunt Group 72: Moving to next stage [7] 2011/10/13 22:11:59: 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=z9hG4bK26d1.376ea0c1.0 Via: SIP/2.0/UDP 83.143.188.161:5060;rport=5060;received=83.143.188.161;branch=z9hG4bK886549626 From: <sip:+31707113070@83.143.188.161;user=phone>;tag=1473184255 To: <sip:31707113070@sip1.budgetphone.nl;user=phone>;tag=1e8fcdff21 Call-ID: 76722836@83.143.188.161 CSeq: 21 BYE Max-Forwards: 12 Reason: Q.850 ;cause=16 ;text="Normal call clearing" Content-Length: 0 [7] 2011/10/13 22:11:59: SIP Tx udp:83.143.188.165:5060: SIP/2.0 200 Ok Via: SIP/2.0/UDP 83.143.188.165;branch=z9hG4bK26d1.376ea0c1.0 Via: SIP/2.0/UDP 83.143.188.161:5060;rport=5060;received=83.143.188.161;branch=z9hG4bK886549626 From: <sip:+31707113070@83.143.188.161;user=phone>;tag=1473184255 To: <sip:31707113070@sip1.budgetphone.nl;user=phone>;tag=1e8fcdff21 Call-ID: 76722836@83.143.188.161 CSeq: 21 BYE Contact: <sip:31707113070@62.148.184.139:5060;transport=udp> User-Agent: snom-PBX/2011-4.2.1.4025 Content-Length: 0 [7] 2011/10/13 22:12:01: SIP Rx udp:83.143.188.165:5060: OPTIONS sip:62.148.184.174:5060 SIP/2.0 Via: SIP/2.0/UDP 83.143.188.165:5060;branch=0 From: sip:pinger@sip1.budgetphone.nl;tag=b2fa69fc To: sip:62.148.184.174:5060 Call-ID: 8e8757a3-9666d35c-881179@83.143.188.165 CSeq: 1 OPTIONS Content-Length: 0 [7] 2011/10/13 22:12:01: SIP Tx udp:83.143.188.165:5060: SIP/2.0 200 Ok Via: SIP/2.0/UDP 83.143.188.165:5060;branch=0 From: <sip:pinger@sip1.budgetphone.nl>;tag=b2fa69fc To: <sip:62.148.184.174:5060>;tag=14868ba08c Call-ID: 8e8757a3-9666d35c-881179@83.143.188.165 CSeq: 1 OPTIONS Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Content-Length: 0

thanks, that seems to work. if somebody knows how to do it through the extensions that would still help (more manageble)

Ok, just tried that but to no avail. Added mobile number to "call forward when not registered" and to "cell phone number" and also set "when calling the extension in a hunt group" to "include cell phone in the group" but nothing happens, just redirects directly to "final stage".

I've setup 9 extensions which redirect all calls to a mobile phone. Now when I use these extensions in a hunt group the mobile phones don't ring but the hunt goes directly to the final stage. Is this possible?

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

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...

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"

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?

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?

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.