Asterisk 1.6 Fax

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 !

first if you post some question about a fax then don’t ask about a sip problem
second please dial the right number when posting a log
you are dialing a 9 digit number instead of the 10 digits that is needed

and what did you do did you upgrade from 1.4 to 1.6 did it ever work?

The dial number is right!
Yes it worked fine with version 1.4. I removed the package asterisk-1.4, downloaded the archive (asterisk-1.6.0.3.tar.gz), installed it, and copied the conf files which I needed(from version 1.4).

Probably. Otherwise calling the number would not result in any log entries.

Did you try to use the config files which came with version 1.6?

I think the SIP problem is what’s preventing him from building his fax system. As long as the inbound call is rejected, he will not be able to test the fax application.

And just applying some guesswork here:

It might be that he upgraded from 1.4 to 1.6 because he wants to use T.38 fax functionality, which is available only in 1.6.

Hearing that he has been copying the config files from 1.4 over, I wonder if there are any new security options which need to be set in 1.6 to make it allow the inbound call.

Please read this:
forums.whirlpool.net.au/forum-re … 33054.html

Regards,
nfh