07-27-2007, 05:00 AM | #1 |
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Join Date: Jul 2007
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SIP Registrations
Hi
what is "SIP Registrations", does it affect me, I just have a VoXaLot basic and do not need webcall back and the forwarding things. thanks |
07-27-2007, 06:10 AM | #2 |
Join Date: Jan 2007
Location: Sydney
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SIP registrations are only required for in-bounds, that is if you want people to be able to call your VSP provided number or you want to set up a Direct Dial-in Number (DID)
Tony. |
07-27-2007, 01:58 PM | #3 |
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I'm trying to call my voxalot accout from another domain but the voxalot server require authorization. Is it normal? I would receive free call from anyone.
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07-27-2007, 02:27 PM | #4 | |
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Join Date: Mar 2006
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change "From" domain
Quote:
The SIP INVITE packet would look like this: Code:
INVITE sip:12xxxx@us.voxalot.com SIP/2.0 Via: SIP/2.0/UDP 221.xx.xx.xx:5060;branch=zh4K500614582 From: 66xxxx <sip:66xxxx@us.voxalot.com>;tag=06810 ... You need to change the "From:" part to something else, like this: Code:
INVITE sip:12xxxx@us.voxalot.com SIP/2.0 Via: SIP/2.0/UDP 221.xx.xx.xx:5060;branch=zh4K500614582 From: 66xxxx <sip:66xxxx@sip.pennytel.com>;tag=06810 ... |
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07-27-2007, 02:56 PM | #5 |
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This is the response I got.
Code:
SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 212.109.160.59:5060;branch=z9hG4bK31c28521;rport=5060 From: "1001" <sip:1001@sip.omnianet.it>;tag=as66afa6f1 To: <sip:190475@voxalot.com>;tag=as3ca8a686 Call-ID: 0cefea390cf655587eb649534aa63d66@sip.omnianet.it CSeq: 103 INVITE Proxy-Authenticate: Digest realm="voxalot.com", nonce="46aa07b5c2f912c19133ae6f638842b98e7712cb" Server: OpenSer (1.1.0-notls (i386/linux)) Content-Length: 0 Warning: 392 64.34.173.199:5060 "Noisy feedback tells: pid=15434 req_src_ip=212.109.160.59 req_src_port=5060 in_uri=sip:190475@64.34.163.35:5061 out_uri=sip:190475@64.34.163.35:5061 via_cnt==1" |
07-27-2007, 05:25 PM | #6 |
Senior Member
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07-28-2007, 12:41 AM | #7 | |
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Join Date: Mar 2006
Posts: 96
Thanks: 8 Thanked 26 Times in 19 Posts |
works for me
Quote:
I've just tried calling my own VoXaLot number, via SIP, and it works for me: SEND >> 64.34.163.35:5060 Code:
INVITE sip:66xxxx@voxalot.com SIP/2.0 Via: SIP/2.0/UDP 221.xx.xx.yyy:5060;branch=z12JUKS83L3KS5 From: 1001 <sip:1001@sip.omnianet.it>;tag=764466127 To: <sip:66xxxx@voxalot.com> Contact: <sip:1001@221.xx.xx.yyy:5060> Call-ID: 98KS32432@221.xx.xx.yyy CSeq: 61224 INVITE Max-Forwards: 70 Content-Type: application/sdp User-Agent: X-Lite release 1103m Content-Length: 194 v=0 o=1001 1687423 1687496 IN IP4 221.xx.xx.yyy s=X-Lite c=IN IP4 221.xx.xx.yyy t=0 0 m=audio 29360 RTP/AVP 0 101 a=rtpmap:0 pcmu/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 RECEIVE << 64.34.163.35:5060 Code:
SIP/2.0 100 Trying Via: SIP/2.0/UDP 221.xx.xx.yyy:5060;branch=z12JUKS83L3KS5 From: 1001 <sip:1001@sip.omnianet.it>;tag=764466127 To: <sip:66xxxx@voxalot.com> Call-ID: 98KS32432@221.xx.xx.yyy CSeq: 61224 INVITE Server: OpenSer (1.1.0-notls (i386/linux)) Content-Length: 0 Warning: 392 64.34.163.35:5060 "Noisy feedback tells: pid=30944 req_src_ip=221.xx.xx.yyy req_src_port=5060 in_uri=sip:66xxxx@voxalot.com out_uri=sip:66xxxx@voxalot.com via_cnt==1" RECEIVE << 64.34.163.35:5060 Code:
SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 221.xx.xx.yyy:5060;branch=z12JUKS83L3KS5 From: 1001 <sip:1001@sip.omnianet.it>;tag=764466127 To: <sip:66xxxx@voxalot.com> Call-ID: 98KS32432@221.xx.xx.yyy CSeq: 61224 INVITE Server: OpenSer (1.1.0-notls (i386/linux)) Content-Length: 0 Warning: 392 64.34.163.35:5060 "Noisy feedback tells: pid=30944 req_src_ip=221.xx.xx.yyy req_src_port=5060 in_uri=sip:66xxxx@voxalot.com out_uri=sip:*01066xxxx@64.34.173.199:5060 via_cnt==1" RECEIVE << 64.34.163.35:5060 Code:
SIP/2.0 180 Ringing Via: SIP/2.0/UDP 221.xx.xx.yyy:5060;branch=z12JUKS83L3KS5 Record-Route: <sip:64.34.173.199;lr=on;ftag=764466127> Record-Route: <sip:64.34.163.35;lr=on;ftag=764466127> From: 1001 <sip:1001@sip.omnianet.it>;tag=764466127 To: <sip:66xxxx@voxalot.com>;tag=1402227574 Call-ID: 98KS32432@221.xx.xx.yyy CSeq: 61224 INVITE Contact: <sip:66xxxx@221.xx.xx.xx:6080> Content-Length: 0 Could you provide more information on your setup? Which VoXaLot server is your SIP device registering with? Have you got call forwarding active? Could you provide a copy of the SIP INVITE message SIP device is sending? |
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07-28-2007, 07:19 AM | #8 |
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Hello. Here is the invite. On this number there is just the voicemail.
Code:
INVITE sip:190475@64.34.163.35:5061 SIP/2.0 Via: SIP/2.0/UDP 212.109.160.59:5060;branch=z9hG4bK2d21db3b;rport Route: <sip:64.34.173.199;lr=on;ftag=as75fcf5be>,<sip:64.34.163.35;lr=on;ftag=as75fcf5be> From: "1001" <sip:1001@omnianet.it>;tag=as75fcf5be To: <sip:190475@voxalot.com>;tag=as2d2dbc73 Contact: <sip:1001@212.109.160.59> Call-ID: 5478b0226d9238c54aaf9ea052283918@omnianet.it CSeq: 103 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 309 v=0 o=root 2200 2201 IN IP4 192.168.10.65 s=session c=IN IP4 192.168.10.65 t=0 0 m=audio 16152 RTP/AVP 0 8 98 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:98 iLBC/8000 a=fmtp:98 mode=30 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv |
07-28-2007, 09:41 AM | #9 |
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Sip registration
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07-28-2007, 09:44 AM | #10 |
Junior Member
Join Date: Jul 2007
Posts: 4
Thanks: 1 Thanked 0 Times in 0 Posts |
Sip registration
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