[Home]

Summary:ASTERISK-14004: Trunk registration / Auth user
Reporter:Mattias (tornblad)Labels:
Date Opened:2009-04-23 00:59:35Date Closed:2009-10-09 13:49:40
Priority:MinorRegression?No
Status:Closed/CompleteComponents:Channels/chan_sip/Registration
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:( 0) 14954.patch
Description:When using Asterisk GUI 2.0 to create a Trunk it adds the following lines to users.conf

[internetcalls.com]
host = sip.internetcalls.com
username = XXXXXXXXXX
secret = XXXXXXXXXX
authuser = XXXXXXXXXXXX
trunkname = internetcalls.com
context = DID_internetcalls.com
group = null
hasexten = no
hasiax = no
hassip = yes
registeriax = no
registersip = yes
trunkstyle = voip
disallow = all
allow = all

but when registering the AUTHUSER parameter is not passed when registering this way. Changing to REGISTERSIP = NO and adding a REGISTER => XXXX:XXXX:XXXX@sip.internetcalls.com works fine.

The same problem in 1.6.1 RCs and in 1.6.0.x

Best solution should be to add all parameters that might be needed....

register => [transport://]user[:secret[:authuser]]@host[:port][/extension]


****** ADDITIONAL INFORMATION ******

.\asterisk-1.6.1.0-rc5\channels\chan_sip.c, starting line 22665


if (ucfg) {
struct ast_variable *gen;
int genhassip, genregistersip;
const char *hassip, *registersip;

genhassip = ast_true(ast_variable_retrieve(ucfg, "general", "hassip"));
genregistersip = ast_true(ast_variable_retrieve(ucfg, "general", "registersip"));
gen = ast_variable_browse(ucfg, "general");
cat = ast_category_browse(ucfg, NULL);
while (cat) {
if (strcasecmp(cat, "general")) {
hassip = ast_variable_retrieve(ucfg, cat, "hassip");
registersip = ast_variable_retrieve(ucfg, cat, "registersip");
if (ast_true(hassip) || (!hassip && genhassip)) {
peer = build_peer(cat, gen, ast_variable_browse(ucfg, cat), 0);
if (peer) {
ao2_t_link(peers, peer, "link peer into peer table");
if ((peer->type & SIP_TYPE_PEER) && peer->addr.sin_addr.s_addr) {
ao2_t_link(peers_by_ip, peer, "link peer into peers_by_ip table");
}

unref_peer(peer, "unref_peer: from reload_config");
peer_count++;
}
}
if (ast_true(registersip) || (!registersip && genregistersip)) {
char tmp[256];
const char *host = ast_variable_retrieve(ucfg, cat, "host");
const char *username = ast_variable_retrieve(ucfg, cat, "username");
const char *secret = ast_variable_retrieve(ucfg, cat, "secret");
const char *contact = ast_variable_retrieve(ucfg, cat, "contact");
if (!host)
host = ast_variable_retrieve(ucfg, "general", "host");
if (!username)
username = ast_variable_retrieve(ucfg, "general", "username");
if (!secret)
secret = ast_variable_retrieve(ucfg, "general", "secret");
if (!contact)
contact = "s";
if (!ast_strlen_zero(username) && !ast_strlen_zero(host)) {
if (!ast_strlen_zero(secret))
snprintf(tmp, sizeof(tmp), "%s:%s@%s/%s", username, secret, host, contact);
else
snprintf(tmp, sizeof(tmp), "%s@%s/%s", username, host, contact);
if (sip_register(tmp, 0) == 0)
registry_count++;
}
}
}
cat = ast_category_browse(ucfg, cat);
}
ast_config_destroy(ucfg);
}

Comments:By: Leif Madsen (lmadsen) 2009-04-27 13:00:41

I *think* this might be a GUI issue, but not positive :)
By: Ryan Brindley (rbrindley) 2009-04-27 15:17:00

Talked with mmichelson,
chan_sip.c was not even reading authuser from users.conf. mmichelson is updating chan_sip to correct this issue.
By: Mark Michelson (mmichelson) 2009-04-27 15:21:13

Give the attached patch a shot.
By: Mattias (tornblad) 2009-05-07 09:16:06

Tried the patch with 1.6.1.0 and it works fine with the register problem, and incoming calls work OK. But I can't make any outgoing calls! I'm not really sure I'm using the parameters USERNAME, AUTHUSER and FROMUSER correct.

My USERS.CONF contains

[Digisip]
type = peer
host = proxy.digisip.net
username = 0812341234
secret = MyPassword
trunkname = Digisip  ; GUI metadata
context = DID_Digisip
group = null
hasexten = no
hasiax = no
hassip = yes
registeriax = no
registersip = yes
trunkstyle = voip
authuser = 1234
insecure = port,invite
fromdomain = proxy.digisip.net
fromuser = 0812341234
disallow = all
allow = alaw,gsm
dtmfmode = rfc2833


And below is some SIP debug info, compare the username in the Register packet, and the Invite packet.


==================== SIP SHOW REGISTRY =====================================
Host                           dnsmgr Username       Refresh State                Reg.Time                
proxy.digisip.net:5060         N      0812341234         105 Registered           Thu, 07 May 2009 12:45:14
sip.internetcalls.com:5060     N      mtornblad          105 Registered           Thu, 07 May 2009 12:45:14

2 SIP registrations.



================== INFO FROM DEBUG LOG WHEN DOING REGISTER ====================

REGISTER 12 headers, 0 lines
Reliably Transmitting (no NAT) to 82.209.165.194:5060:
REGISTER sip:proxy.digisip.net SIP/2.0
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK034d7d61;rport
Max-Forwards: 70
From: <sip:0812341234@proxy.digisip.net>;tag=as3f1a271f
To: <sip:0812341234@proxy.digisip.net>
Call-ID: 6902a10b4e153f22244dbb87379680fb@brittatorp.se
CSeq: 105 REGISTER
User-Agent: Asterisk PBX
Authorization: Digest username="1234", realm="proxy.digisip.net", algorithm=MD5, uri="sip:proxy.digisip.net", nonce="4a02bd9bfe581703efc503733b2bf94d65592b5a", response="4472cb5782854d4e876355b3664a2477"
Expires: 120
Contact: <sip:s@192.168.1.11>
Content-Length: 0


<--- SIP read from UDP://82.209.165.194:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK034d7d61;rport=63006
From: <sip:0812341234@proxy.digisip.net>;tag=as3f1a271f
To: <sip:0812341234@proxy.digisip.net>
Call-ID: 6902a10b4e153f22244dbb87379680fb@brittatorp.se
CSeq: 105 REGISTER
Server: Sip EXpress router (0.9.3 (i386/linux))
Content-Length: 0
Warning: 392 82.209.165.194:5060 "Noisy feedback tells:  pid=7781 req_src_ip=192.168.1.11 req_src_port=63006 in_uri=sip:proxy.digisip.net out_uri=sip:proxy.digisip.net via_cnt==1"


<--- SIP read from UDP://82.209.165.194:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK034d7d61;rport=63006
From: <sip:0812341234@proxy.digisip.net>;tag=as3f1a271f
To: <sip:0812341234@proxy.digisip.net>;tag=9d0c2f86226119ed3517373c02465a4b.17cc
Call-ID: 6902a10b4e153f22244dbb87379680fb@brittatorp.se
CSeq: 105 REGISTER
Contact: <sip:s@192.168.1.11:5060>;expires=120
Server: Sip EXpress router (0.9.3 (i386/linux))
Content-Length: 0
Warning: 392 82.209.165.194:5060 "Noisy feedback tells:  pid=7781 req_src_ip=192.168.1.11 req_src_port=63006 in_uri=sip:proxy.digisip.net out_uri=sip:proxy.digisip.net via_cnt==1"


============= INFO FROM DEBUG LOG WHEN DOING INVITE ==========================

Reliably Transmitting (no NAT) to 82.209.165.194:5060:
INVITE sip:047012345@proxy.digisip.net SIP/2.0
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK222680d3;rport
Max-Forwards: 70
From: "asterisk" <sip:0812341234@proxy.digisip.net>;tag=as539f2c86
To: <sip:047012345@proxy.digisip.net>
Contact: <sip:0812341234@192.168.1.11>
Call-ID: 567e526c1acad88a57bc7f5503cd03da@proxy.digisip.net
CSeq: 105 INVITE
User-Agent: Asterisk PBX
Proxy-Authorization: Digest username="0812341234", realm="proxy.digisip.net", algorithm=MD5, uri="sip:047012345@proxy.digisip.net", nonce="4a02c6abcba94a4239a1e4d820e9a9bca2f990b0", response="80eafbbfb82dc882f58e6c50a4298584"
Date: Thu, 07 May 2009 11:23:55 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 284

v=0
o=root 315638187 315638190 IN IP4 192.168.1.11
s=Asterisk PBX 1.6.1.0
c=IN IP4 192.168.1.11
t=0 0
m=audio 11704 RTP/AVP 8 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv


<--- SIP read from UDP://82.209.165.194:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK222680d3;rport=63006
From: "asterisk" <sip:0812341234@proxy.digisip.net>;tag=as539f2c86
To: <sip:047012345@proxy.digisip.net>;tag=9d0c2f86226119ed3517373c02465a4b.5f2b
Call-ID: 567e526c1acad88a57bc7f5503cd03da@proxy.digisip.net
CSeq: 105 INVITE
Proxy-Authenticate: Digest realm="proxy.digisip.net", nonce="4a02c6abcba94a4239a1e4d820e9a9bca2f990b0"
Server: Sip EXpress router (0.9.3 (i386/linux))
Content-Length: 0
Warning: 392 82.209.165.194:5060 "Noisy feedback tells:  pid=7793 req_src_ip=192.168.1.11 req_src_port=63006 in_uri=sip:047012345@proxy.digisip.net out_uri=sip:047012345@proxy.digisip.net via_cnt==1"


Transmitting (no NAT) to 82.209.165.194:5060:
ACK sip:047012345@proxy.digisip.net SIP/2.0
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK222680d3;rport
Max-Forwards: 70
From: "asterisk" <sip:0812341234@proxy.digisip.net>;tag=as539f2c86
To: <sip:047012345@proxy.digisip.net>;tag=9d0c2f86226119ed3517373c02465a4b.5f2b
Contact: <sip:0812341234@192.168.1.11>
Call-ID: 567e526c1acad88a57bc7f5503cd03da@proxy.digisip.net
CSeq: 105 ACK
User-Agent: Asterisk PBX
Content-Length: 0


[May  7 13:23:55] NOTICE[23683]: chan_sip.c:16048 handle_response_invite: Failed to authenticate on INVITE to '"asterisk" <sip:0812341234@proxy.digisip.net>;tag=as539f2c86'




I can overide the authentication problem and make outgoing calls work by updating SIP.CONF with ....

auth = 1234:MyPassword@proxy.digisip.net
By: David Vossel (dvossel) 2009-10-09 12:52:09

mmichelson's patch appears to addresses the registration issue correctly, unless you feel the problem you are having with outgoing calls is directly related to this patch please create separate issue for it.  Thanks!
By: Digium Subversion (svnbot) 2009-10-09 12:56:03

Repository: asterisk
Revision: 223205

U   branches/1.4/channels/chan_sip.c

------------------------------------------------------------------------
r223205 | dvossel | 2009-10-09 12:56:03 -0500 (Fri, 09 Oct 2009) | 10 lines

(closes issue ASTERISK-14004)
Reported by: tornblad
Tested by: mmichelson, tornblad, dvossel
fixes sip registration using authuser in user.conf

(closes issue ASTERISK-14004)
Reported by: tornblad
Tested by: mmichelson, tornblad, dvossel


------------------------------------------------------------------------

http://svn.digium.com/view/asterisk?view=rev&revision=223205
By: Digium Subversion (svnbot) 2009-10-09 12:57:05

Repository: asterisk
Revision: 223206

_U  trunk/
U   trunk/channels/chan_sip.c

------------------------------------------------------------------------
r223206 | dvossel | 2009-10-09 12:57:05 -0500 (Fri, 09 Oct 2009) | 16 lines

Merged revisions 223205 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
 r223205 | dvossel | 2009-10-09 12:52:35 -0500 (Fri, 09 Oct 2009) | 10 lines
 
 (closes issue ASTERISK-14004)
 Reported by: tornblad
 Tested by: mmichelson, tornblad, dvossel
 fixes sip registration using authuser in user.conf
 
 (closes issue ASTERISK-14004)
 Reported by: tornblad
 Tested by: mmichelson, tornblad, dvossel
........

------------------------------------------------------------------------

http://svn.digium.com/view/asterisk?view=rev&revision=223206
By: Digium Subversion (svnbot) 2009-10-09 12:59:18

Repository: asterisk
Revision: 223208

_U  branches/1.6.2/
U   branches/1.6.2/channels/chan_sip.c

------------------------------------------------------------------------
r223208 | dvossel | 2009-10-09 12:59:18 -0500 (Fri, 09 Oct 2009) | 23 lines

Merged revisions 223206 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
 r223206 | dvossel | 2009-10-09 12:53:37 -0500 (Fri, 09 Oct 2009) | 16 lines
 
 Merged revisions 223205 via svnmerge from
 https://origsvn.digium.com/svn/asterisk/branches/1.4
 
 ........
   r223205 | dvossel | 2009-10-09 12:52:35 -0500 (Fri, 09 Oct 2009) | 10 lines
   
   (closes issue ASTERISK-14004)
   Reported by: tornblad
   Tested by: mmichelson, tornblad, dvossel
   fixes sip registration using authuser in user.conf
   
   (closes issue ASTERISK-14004)
   Reported by: tornblad
   Tested by: mmichelson, tornblad, dvossel
 ........
................

------------------------------------------------------------------------

http://svn.digium.com/view/asterisk?view=rev&revision=223208
By: Digium Subversion (svnbot) 2009-10-09 12:59:53

Repository: asterisk
Revision: 223209

_U  branches/1.6.1/
U   branches/1.6.1/channels/chan_sip.c

------------------------------------------------------------------------
r223209 | dvossel | 2009-10-09 12:59:53 -0500 (Fri, 09 Oct 2009) | 23 lines

Merged revisions 223206 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
 r223206 | dvossel | 2009-10-09 12:53:37 -0500 (Fri, 09 Oct 2009) | 16 lines
 
 Merged revisions 223205 via svnmerge from
 https://origsvn.digium.com/svn/asterisk/branches/1.4
 
 ........
   r223205 | dvossel | 2009-10-09 12:52:35 -0500 (Fri, 09 Oct 2009) | 10 lines
   
   (closes issue ASTERISK-14004)
   Reported by: tornblad
   Tested by: mmichelson, tornblad, dvossel
   fixes sip registration using authuser in user.conf
   
   (closes issue ASTERISK-14004)
   Reported by: tornblad
   Tested by: mmichelson, tornblad, dvossel
 ........
................

------------------------------------------------------------------------

http://svn.digium.com/view/asterisk?view=rev&revision=223209
By: Digium Subversion (svnbot) 2009-10-09 13:00:32

Repository: asterisk
Revision: 223210

_U  branches/1.6.0/
U   branches/1.6.0/channels/chan_sip.c

------------------------------------------------------------------------
r223210 | dvossel | 2009-10-09 13:00:31 -0500 (Fri, 09 Oct 2009) | 23 lines

Merged revisions 223206 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
 r223206 | dvossel | 2009-10-09 12:53:37 -0500 (Fri, 09 Oct 2009) | 16 lines
 
 Merged revisions 223205 via svnmerge from
 https://origsvn.digium.com/svn/asterisk/branches/1.4
 
 ........
   r223205 | dvossel | 2009-10-09 12:52:35 -0500 (Fri, 09 Oct 2009) | 10 lines
   
   (closes issue ASTERISK-14004)
   Reported by: tornblad
   Tested by: mmichelson, tornblad, dvossel
   fixes sip registration using authuser in user.conf
   
   (closes issue ASTERISK-14004)
   Reported by: tornblad
   Tested by: mmichelson, tornblad, dvossel
 ........
................

------------------------------------------------------------------------

http://svn.digium.com/view/asterisk?view=rev&revision=223210
By: Digium Subversion (svnbot) 2009-10-09 13:47:03

Repository: asterisk
Revision: 223206

_U  trunk/
U   trunk/channels/chan_sip.c

------------------------------------------------------------------------
r223206 | dvossel | 2009-10-09 12:53:37 -0500 (Fri, 09 Oct 2009) | 13 lines

Merged revisions 223205 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
 r223205 | dvossel | 2009-10-09 12:52:35 -0500 (Fri, 09 Oct 2009) | 10 lines
 
 fixes sip registration using authuser in user.conf
 
 (closes issue ASTERISK-14004)
 Reported by: tornblad
 Tested by: mmichelson, tornblad, dvossel
........

------------------------------------------------------------------------

http://svn.digium.com/view/asterisk?view=rev&revision=223206
By: Digium Subversion (svnbot) 2009-10-09 13:47:59

Repository: asterisk
Revision: 223205

U   branches/1.4/channels/chan_sip.c

------------------------------------------------------------------------
r223205 | dvossel | 2009-10-09 12:52:35 -0500 (Fri, 09 Oct 2009) | 8 lines


fixes sip registration using authuser in user.conf

(closes issue ASTERISK-14004)
Reported by: tornblad
Tested by: mmichelson, tornblad, dvossel


------------------------------------------------------------------------

http://svn.digium.com/view/asterisk?view=rev&revision=223205
By: Digium Subversion (svnbot) 2009-10-09 13:48:43

Repository: asterisk
Revision: 223208

_U  branches/1.6.2/
U   branches/1.6.2/channels/chan_sip.c

------------------------------------------------------------------------
r223208 | dvossel | 2009-10-09 12:55:50 -0500 (Fri, 09 Oct 2009) | 20 lines

Merged revisions 223206 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
 r223206 | dvossel | 2009-10-09 12:53:37 -0500 (Fri, 09 Oct 2009) | 16 lines
 
 Merged revisions 223205 via svnmerge from
 https://origsvn.digium.com/svn/asterisk/branches/1.4
 
 ........
   r223205 | dvossel | 2009-10-09 12:52:35 -0500 (Fri, 09 Oct 2009) | 10 lines
   
   fixes sip registration using authuser in user.conf
   
(closes issue ASTERISK-14004)
   Reported by: tornblad
   Tested by: mmichelson, tornblad, dvossel
 ........
................

------------------------------------------------------------------------

http://svn.digium.com/view/asterisk?view=rev&revision=223208
By: Digium Subversion (svnbot) 2009-10-09 13:49:11

Repository: asterisk
Revision: 223209

_U  branches/1.6.1/
U   branches/1.6.1/channels/chan_sip.c

------------------------------------------------------------------------
r223209 | dvossel | 2009-10-09 12:56:26 -0500 (Fri, 09 Oct 2009) | 19 lines

Merged revisions 223206 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
 r223206 | dvossel | 2009-10-09 12:53:37 -0500 (Fri, 09 Oct 2009) | 16 lines
 
 Merged revisions 223205 via svnmerge from
 https://origsvn.digium.com/svn/asterisk/branches/1.4
 
 ........
   r223205 | dvossel | 2009-10-09 12:52:35 -0500 (Fri, 09 Oct 2009) | 10 lines
   fixes sip registration using authuser in user.conf
   
   (closes issue ASTERISK-14004)
   Reported by: tornblad
   Tested by: mmichelson, tornblad, dvossel
 ........
................

------------------------------------------------------------------------

http://svn.digium.com/view/asterisk?view=rev&revision=223209
By: Digium Subversion (svnbot) 2009-10-09 13:49:39

Repository: asterisk
Revision: 223210

_U  branches/1.6.0/
U   branches/1.6.0/channels/chan_sip.c

------------------------------------------------------------------------
r223210 | dvossel | 2009-10-09 12:57:04 -0500 (Fri, 09 Oct 2009) | 19 lines

Merged revisions 223206 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
 r223206 | dvossel | 2009-10-09 12:53:37 -0500 (Fri, 09 Oct 2009) | 16 lines
 
 Merged revisions 223205 via svnmerge from
 https://origsvn.digium.com/svn/asterisk/branches/1.4
 
 ........
   r223205 | dvossel | 2009-10-09 12:52:35 -0500 (Fri, 09 Oct 2009) | 10 lines
   fixes sip registration using authuser in user.conf
   
   (closes issue ASTERISK-14004)
   Reported by: tornblad
   Tested by: mmichelson, tornblad, dvossel
 ........
................

------------------------------------------------------------------------

http://svn.digium.com/view/asterisk?view=rev&revision=223210