Asterisk Error 403 Forbidden
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the gsm incoming calls to this extension. The IP of the sip 403 forbidden asterisk adapter is 192.168.100.120. The IP of Asterisk is 192.168.100.4. My error 403 forbidden fix configuration is this: Peer details:context=from-trunkhost=192.168.100.120qualify=yesnat=notype=peerinsecure=invitedisallow=allallow=ulawusername=2001secret=xxxx Register string: 2001:xxxx@192.168.100.120 Outgoin calls work ok, but incomming calls are error 403 forbidden you don't have permission to access rejected by asterisk with the following error: SIP/2.0 403 ForbiddenVia: SIP/2.0/UDP 192.168.100.120:5060;branch=z9hG4bK28902eff;received=192.168.100.120;rport=5060From: "5045" ;tag=OneStream4534ad57To: ;tag=as3d780b27Call-ID: 57d99a664c11126d6329a0c430ed6e45@192.168.100.120CSeq: 103 INVITEServer: FPBX-2.9.0(1.6.2.18.2)Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
Error 403 Forbidden Apache
NOTIFY, INFOSupported: replaces, timerContent-Length: 0 Any help please? sanjayws 2012-05-17 04:05:49 UTC #2 Shouldn;t the type be user or friend? for incoming...? anv 2012-05-17 05:56:02 UTC #3 I've tried with friend. The same problem. It seems that asteisk is rejecting the from.... but the type=peer or friend should allow any cid, isn't error 403 forbidden mac it? obelisk 2012-05-17 06:03:02 UTC #4 what does asterisk log when the call is rejected ? turn sip trace off anv 2012-05-17 18:36:22 UTC #5 I found the problem. I am calling from a short-number phone. This number is equal to the number of an extension. Then, the call goas from 5045@192.168.100.120 to s@192.168.100.4 but there is an 5045 extension at 192.168.100.4 It seems that Asterisk is rejecting the call because a 5045@192.168.100.104... I don' t know if there is a solution for this other than patching the number at the gsm gateway... obelisk 2012-05-17 21:42:57 UTC #6 change the type of the extension 5045 to be peer instead of friend. I do not know how many times this issue needs to bite people in the a** before the default is changed:http://www.freepbx.org/trac/ticket/5309 system (system) 2014-06-04 19:34:46 UTC #7 Home Categories FAQ/Guidelines Terms of Service Privacy Policy Powered by Discourse, best viewed with JavaScript enabled
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Help Receive Real-Time Help Create a Freelance Project Hire for a Full Time Job Ways to Get Help Ask a Question Ask for Help Receive Real-Time Help Create a Freelance Project Hire for a Full https://www.experts-exchange.com/questions/27239145/xlite-works-asterisk-returns-403-error.html Time Job Ways to Get Help Expand Search Submit Close Search Login Join Today Products BackProducts Gigs Live Careers Vendor Services Groups Website Testing Store Headlines Experts Exchange > Questions > xlite works, http://www.geekzone.co.nz/forums.asp?forumid=43&topicid=157683 asterisk returns "403" error Want to Advertise Here? Solved xlite works, asterisk returns "403" error Posted on 2011-08-04 IP Telephony Voice Over IP 5 Verified Solutions 27 Comments 2,201 Views Last Modified: 2013-11-12 403 forbidden Hi all: Has anyone here tried using indigonetworks' SIP line with Asterisk? There SIP account worked fine with Xlite. And I was able to register SIP account with Asterisk. However, when I try to make calls through these trunks, all I get was the 403 forbidden message back from the provider. Nothing much informational. And believe me that I have tried every combination of trunk error 403 forbidden settings there would be trying to dial out with these SIP lines. The asterisk server sits behind a sonicwall firewall. I was able to make phone calls by using vitelity and other SIP providers. However i need to use indigo's trunk - the other providers were used just for testing. Here is a copy of the log: -- Executing [s@macro-dialout-trunk:19] Dial("SIP/705-0000007f", "SIP/2420000000-out/12423273792,300,") in new stack == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 Audio is at 5060 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x1000 (g722) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 69.4.164.10:5060: INVITE sip:12423273792@nas-sbc-01.srg.com.bs;user=phone SIP/2.0 Via: SIP/2.0/UDP MY-IP:5060;branch=z9hG4bK231d0451;rport Max-Forwards: 70 From: "2420000000"
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