Hi everyone,
First - sorry for my bad English
I have had a problem with asterisk 1.6. When I configured it and after that installed, I haven’t had the problem.
I add in sip.conf my remote sip.
> h03-v04:/etc/asterisk# asterisk -r
>
> h03-v04*CLI> sip show registry
> Host Username Refresh State
> Reg.Time
> 212.50.11.4:5060 030154595 105 Registered
> Tue, 20 Jan 2009 09:43:12
> 1 SIP registrations.
> h03-v04*CLI>
>
It is registered, but when I tried to call I was rejected.
> h03-v04*CLI> sip set debug on
> SIP Debugging enabled
> h03-v04*CLI>
> <--- SIP read from [UDP://212.50.11.4:5060](UDP://212.50.11.4:5060) --->
> INVITE sip:s@78.47.186.84 SIP/2.0
> Via: SIP/2.0/UDP
> 212.50.11.4:5060;branch=z9hG4bK-ff5f140056ff7510ff00001517ffff56
> From:
> ;tag=ff5e140056ff7510ff00001517ffff56
> To:
> Call-ID: 245e1400569d751080000015178dab56@voip-gw-bg
> CSeq: 1 INVITE
> Contact:
> Max-Forwards: 10
> User-Agent: MERA MSIP v.1.0.2
> Cisco-Guid: 1307077136-3886027229-2326528003-3126573561
> Content-Type: application/sdp
> Content-Length: 238
>
> v=0
> o=- 1232444758 1232444758 IN IP4 212.50.11.4
> s=-
> c=IN IP4 212.50.11.4
> t=0 0
> m=audio 21008 RTP/AVP 18 4 8 101
> a=rtpmap:18 G729/8000
> a=rtpmap:4 G723/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
>
> <------------->
> --- (12 headers 11 lines) ---
> Sending to 212.50.11.4 : 5060 (no NAT)
> Using INVITE request as basis request -
> 245e1400569d751080000015178dab56@voip-gw-bg
> No user '0897897328' in SIP users list
> Found peer 'spnet' for '0897897328' from 212.50.11.4:5060
> h03-v04*CLI>
> <--- Reliably Transmitting (no NAT) to 212.50.11.4:5060 --->
> SIP/2.0 401 Unauthorized
> Via: SIP/2.0/UDP
> 212.50.11.4:5060;branch=z9hG4bK-ff5f140056ff7510ff00001517ffff56;received=212.50.11.4
> From:
> ;tag=ff5e140056ff7510ff00001517ffff56
> To: ;tag=as193ad513
> Call-ID: 245e1400569d751080000015178dab56@voip-gw-bg
> CSeq: 1 INVITE
> User-Agent: Asterisk PBX 1.6.0.3-rc1
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces, timer
> WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0a366714"
> Content-Length: 0
>
>
> <------------>
> Scheduling destruction of SIP dialog
> '245e1400569d751080000015178dab56@voip-gw-bg' in 32000 ms (Method: INVITE)
> h03-v04*CLI>
> <--- SIP read from [UDP://212.50.11.4:5060](UDP://212.50.11.4:5060) --->
> ACK sip:030154595@78.47.186.84:5060;user=phone SIP/2.0
> Via: SIP/2.0/UDP
> 212.50.11.4:5060;branch=z9hG4bK-ff5f140056ff7510ff00001517ffff56;received=212.50.11.4
> From:
> ;tag=ff5e140056ff7510ff00001517ffff56
> To: ;tag=as193ad513
> Call-ID: 245e1400569d751080000015178dab56@voip-gw-bg
> CSeq: 1 ACK
> Max-Forwards: 10
> User-Agent: MERA MSIP v.1.0.2
> Content-Length: 0
>
>
> <------------->
> --- (9 headers 0 lines) ---
> h03-v04*CLI>
I use debian linux, and I don’t use the packet system for install asterisk, because I need to make it a fax system !
btw: if anyone has already configured files for a fax system with asterisk, please to send me them.
Thank you !