403 Forbidden Error Sip
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SIP is based around request/response transactions, in a similar manner to the Hypertext Transfer Protocol (HTTP). Each transaction consists of a SIP request (which
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will be one of several request methods), and at least one sip 403 forbidden error asterisk response.[1]:p11 SIP requests and responses may be generated by any SIP user agent; user agents are divided into
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clients (UACs), which initiate requests, and servers (UASes), which respond to them.[1]:§8 A single user agent may act as both UAC and UAS for different transactions:[1]:p26 for example, a sip 488 SIP phone is a user agent that will be a UAC when making a call, and a UAS when receiving one. Additionally, some devices will act as both UAC and UAS for a single transaction; these are called Back-to-Back User Agents (B2BUAs).[1]:p20 SIP responses specify a three-digit integer response code, which is one of a number of defined sip 422 codes that detail the status of the request. These codes are grouped according to their first digit as "provisional", "success", "redirection", "client error", "server error" or "global failure" codes, corresponding to a first digit of 1–6; these are expressed as, for example, "1xx" for provisional responses with a code of 100–199.[1]:§7.2 The SIP response codes are an extension to the HTTP response codes, although not all HTTP response codes are valid in SIP.[1]:§21 SIP responses also specify a "reason phrase", and a default reason phrase is defined with each response code.[1]:§7.2 These reason phrases can be varied, however, such as to provide additional information[1]:§21.4.18 or to provide the text in a different language.[1]:§20.3 SIP, including the response codes and corresponding reason phrases, is defined in Internet Engineering Task Force (IETF) Requests for Comments (RFCs), namely RFC 3261.[2] That RFC includes provisions for later RFCs to update the specification.[1]:§8.1.1.9 Specific parts of the SIP protocol, including response codes and their default reason phrases, are registered with the Internet Assigned Numbers Authority (IANA).[1]:§27[3] T
the gsm incoming calls to this extension. The IP of the
Sip Status Codes
adapter is 192.168.100.120. The IP of Asterisk is 192.168.100.4. My sip 403 forbidden asterisk 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
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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, https://en.wikipedia.org/wiki/List_of_SIP_response_codes 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 http://community.freepbx.org/t/403-forbidden-when-receiving-from-sip-trunk/14194 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
to my SIP http://forum.yate.ro/index.php?topic=121.0 provider, get 403 forbidden « previous next » Print Pages: [1] Author Topic: Can't register to my SIP provider, get 403 forbidden http://www.sip-ua.com/wiki/index.php/SIP/2.0_403_Forbidden_error (Read 11748 times) terminus Newbie Posts: 2 Can't register to my SIP provider, get 403 forbidden « on: January 26, 2013, 403 forbidden 09:36:29 PM » I have read the documentation but I am still having trouble making a call through my SIP provider. In this instance I'm trying to dial a test number "775" from a handset called "jeremy". Here are the relevant parts sip 403 forbidden of my files (IP addresses and passwords changed to protect the innocent):accfile.conf:Code: [Select][pennytel]
enabled=yes
protocol=sip
username=8881235678
password=providerpassword
description=Pennytel
registrar=sip.pennytel.comregexroute.conf:Code: [Select]${username}^$=-;error=noauth
^\(7..\)=sip/\1;line=pennytelregfile.conf:Code: [Select][jeremy]
password=userpasswordAnd here is the relevant part of the log:Code: [Select]#
U 85.234.150.65:5060 -> 202.85.243.115:5060
INVITE sip:775@sip.pennytel.com SIP/2.0.
Max-Forwards: 19.
Via: SIP/2.0/UDP 85.234.150.65:5060;rport;branch=z9hG4bK953691761.
From: "Jeremy Malcolm"
To:
Call-ID: 1402492972@sip.pennytel.com.
CSeq: 5 INVITE.
User-Agent: YATE/2.2.0.
Contact:
Allow: ACK, INVITE, BYE, CANCEL, REGISTER, REFER, OPTIONS, INFO.
Content-Type: application/sdp.
Content-Length: 232.
.
v=0.
o=yate 1359255461 1359255461 IN IP4 85.234.150.65.
s=SIP Call.
c=IN IP4 85.234.150.65.
t=0 0.
m=audio 26276 RTP/AVP 3 0 8 101.
a=rtpmap:3 GSM/8000.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
#
U 202.85.243.115:5060 -> 85.234.150.65:5060
SIP/2.0 401 Unauthorized.
Via: SIP/2.0/UDP 85.234.150.65:5060;rport=5060;branch=z9hG4bK953691761.
From: "Jeremy Malcolm"
To:
Call-ID: 1402492972@sip.pennytel.com.
CSeq: 5 INVITE.
the SIP server you are registered to, there is a good chance your router is blocking the incoming call due to the toll-faud prevention feature that was added to IOS version 15.1(2)T. Click Here to learn how to configure your router to allow SIP connections from the SIP-UA.com servers. Retrieved from "http://www.sip-ua.com/wiki/index.php/SIP/2.0_403_Forbidden_error" What links here Related changes Special pages Printable version Permanent link This page was last modified on 22 April 2011, at 17:02. This page has been accessed 6,878 times. Privacy policy About Help Disclaimers Powered by MediaWiki Designed by Paul Gu