Re: [asterisk-users] Authentication Issue Between Servers
No one ever responded to this inquiry but I figured out what the issue
was. I thought I would respond with the solution just in case someone
runs into the same issue in the future.
Firstly, when setting up trunking between servers the "username =" field
is not optional. :) Also, I had a lot of extra fields in place that I
didn't need but hadn't taken the time to remove. I have developed the
opinion that config files should be kept as lean as possible. Here is
the revised SIP peer configuration from sip.conf:
[trunk]
type = friend
username = trunk
callerid =
context = default
host = 172.21.235.1
secret = password
canreinvite = no
disallow = all
allow = gsm
Joshua Billings wrote:
I've got an issue where I am trying to route calls between Asterisk
Servers. I can route calls inbound to a server but seem to have an
authentication issue going out over the same sip account. It appears
that my server isn't sending the second invite after proxy
authentication request. I can't figure out why; any ideas would be
greatly appreciated. Thanks!
- Josh
Here is my sip.conf:
[general]
context = default
allowoverlap = no
bindport = 5060
bindaddr = 0.0.0.0
srvlookup = yes
externip = 172.21.235.2
localnet = 172.21.235.2/255.255.0.0
dtmfmode = rfc2833
relaxdtmf = yes
disallow = all
allow = ulaw
allow = gsm
maxexpirey = 30
defaultexpirey = 180
relaxdtmf=yes
canreinvite = no
nat = 0
UserAgent = Asterisk
echocancel = yes
echocancelwhenbridge = yes
t38pt_udptl = no
[trunk]
type = friend
callwaiting = yes
caller id =
contact =
context = default
fullname =
group =
hasagent = no
hasdirectory = yes
hasiax = no
hasmanager = no
hassip = yes
host = 172.21.235.1
secret = [password]
threewaycalling = yes
registersip = yes
canreinvite = no
nat = no
dtmfmode = rfc2833
registeriax = no
disallow = all
allow = gsm
register=>trunk:[passwo...@172.21.235.1
Here is the applicable portion of extensions.conf:
[default]
exten = _5XX,1,Dial(SIP/trunk/${EXTEN},,Tt)
Here is the SIP Debug output:
INVITE sip:5...@172.21.235.1 SIP/2.0
Via: SIP/2.0/UDP 172.21.235.2:5060;branch=z9hG4bK78d4e8d7;rport
From: "Marci" ;tag=as5951033c
To:
Contact:
Call-ID: 430c49156ce4a7500b1fa57807b5a...@172.21.235.2
CSeq: 102 INVITE
User-Agent: Asterisk
Max-Forwards: 70
Date: Tue, 30 Jun 2009 19:09:25 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 239
v=0
o=root 11411 11411 IN IP4 172.21.235.2
s=session
c=IN IP4 172.21.235.2
t=0 0
m=audio 11486 RTP/AVP 3 101
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
^@
^[[KWBPBXFG000304*CLI>
<--- SIP read from 172.21.235.1:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
172.21.235.2:5060;branch=z9hG4bK78d4e8d7;received=172.21.235.2;rport=5060
From: "Marci" ;tag=as5951033c
To: ;tag=as045cd609
Call-ID: 430c49156ce4a7500b1fa57807b5a...@172.21.235.2
CSeq: 102 INVITE
User-Agent: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk",
nonce="4c4374da"
Content-Length: 0
<->
^@
^[[KWBPBXFG000304*CLI>
--- (11 headers 0 lines) ---
^@
^[[KWBPBXFG000304*CLI>
Transmitting (NAT) to 172.21.235.1:5060:
ACK sip:5...@172.21.235.1 SIP/2.0
Via: SIP/2.0/UDP 172.21.235.2:5060;branch=z9hG4bK78d4e8d7;rport
From: "Marci" ;tag=as5951033c
To: ;tag=as045cd609
Contact:
Call-ID: 430c49156ce4a7500b1fa57807b5a...@172.21.235.2
CSeq: 102 ACK
User-Agent: Asterisk
Max-Forwards: 70
Content-Length: 0
---
^@
^[[KWBPBXFG000304*CLI>
[Jun 30 14:09:25] NOTICE[11434]: chan_sip.c:12253
handle_response_invite: ^...@failed to authenticate on INVITE to '"Marci"
;tag=as5951033c'
^@
^[[KWBPBXFG000304*CLI>
Really destroying SIP dialog
'430c49156ce4a7500b1fa57807b5a...@172.21.235.2' Method: INVITE
^@
^[[KWBPBXFG000304*CLI>
Really destroying SIP dialog
'0fe5f50f7674160d2ab3522f09060...@127.0.0.1' Method: REGISTER
^@
^[[KWBPBXFG000304*CLI>
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[asterisk-users] Authentication Issue Between Servers
I've got an issue where I am trying to route calls between Asterisk
Servers. I can route calls inbound to a server but seem to have an
authentication issue going out over the same sip account. It appears
that my server isn't sending the second invite after proxy
authentication request. I can't figure out why; any ideas would be
greatly appreciated. Thanks!
- Josh
Here is my sip.conf:
[general]
context = default
allowoverlap = no
bindport = 5060
bindaddr = 0.0.0.0
srvlookup = yes
externip = 172.21.235.2
localnet = 172.21.235.2/255.255.0.0
dtmfmode = rfc2833
relaxdtmf = yes
disallow = all
allow = ulaw
allow = gsm
maxexpirey = 30
defaultexpirey = 180
relaxdtmf=yes
canreinvite = no
nat = 0
UserAgent = Asterisk
echocancel = yes
echocancelwhenbridge = yes
t38pt_udptl = no
[trunk]
type = friend
callwaiting = yes
caller id =
contact =
context = default
fullname =
group =
hasagent = no
hasdirectory = yes
hasiax = no
hasmanager = no
hassip = yes
host = 172.21.235.1
secret = [password]
threewaycalling = yes
registersip = yes
canreinvite = no
nat = no
dtmfmode = rfc2833
registeriax = no
disallow = all
allow = gsm
register=>trunk:[passwo...@172.21.235.1
Here is the applicable portion of extensions.conf:
[default]
exten = _5XX,1,Dial(SIP/trunk/${EXTEN},,Tt)
Here is the SIP Debug output:
INVITE sip:5...@172.21.235.1 SIP/2.0
Via: SIP/2.0/UDP 172.21.235.2:5060;branch=z9hG4bK78d4e8d7;rport
From: "Marci" ;tag=as5951033c
To:
Contact:
Call-ID: 430c49156ce4a7500b1fa57807b5a...@172.21.235.2
CSeq: 102 INVITE
User-Agent: Asterisk
Max-Forwards: 70
Date: Tue, 30 Jun 2009 19:09:25 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 239
v=0
o=root 11411 11411 IN IP4 172.21.235.2
s=session
c=IN IP4 172.21.235.2
t=0 0
m=audio 11486 RTP/AVP 3 101
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
^@
^[[KWBPBXFG000304*CLI>
<--- SIP read from 172.21.235.1:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
172.21.235.2:5060;branch=z9hG4bK78d4e8d7;received=172.21.235.2;rport=5060
From: "Marci" ;tag=as5951033c
To: ;tag=as045cd609
Call-ID: 430c49156ce4a7500b1fa57807b5a...@172.21.235.2
CSeq: 102 INVITE
User-Agent: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4c4374da"
Content-Length: 0
<->
^@
^[[KWBPBXFG000304*CLI>
--- (11 headers 0 lines) ---
^@
^[[KWBPBXFG000304*CLI>
Transmitting (NAT) to 172.21.235.1:5060:
ACK sip:5...@172.21.235.1 SIP/2.0
Via: SIP/2.0/UDP 172.21.235.2:5060;branch=z9hG4bK78d4e8d7;rport
From: "Marci" ;tag=as5951033c
To: ;tag=as045cd609
Contact:
Call-ID: 430c49156ce4a7500b1fa57807b5a...@172.21.235.2
CSeq: 102 ACK
User-Agent: Asterisk
Max-Forwards: 70
Content-Length: 0
---
^@
^[[KWBPBXFG000304*CLI>
[Jun 30 14:09:25] NOTICE[11434]: chan_sip.c:12253
handle_response_invite: ^...@failed to authenticate on INVITE to '"Marci"
;tag=as5951033c'
^@
^[[KWBPBXFG000304*CLI>
Really destroying SIP dialog
'430c49156ce4a7500b1fa57807b5a...@172.21.235.2' Method: INVITE
^@
^[[KWBPBXFG000304*CLI>
Really destroying SIP dialog
'0fe5f50f7674160d2ab3522f09060...@127.0.0.1' Method: REGISTER
^@
^[[KWBPBXFG000304*CLI>
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Re: [asterisk-users] authentication issue!
You may use astdb for this.
Just set an entry on AstDB with user password and then for every outgoing
call prompt an audio to introduce password and then check if it exists on
AstDB.
User may be the caller ID and the pass is introduced by DTMF.
Then you may use a GOTOIF to allow or not Outgoing Call.
Other choice could be easier:
There are other ways with Authenticate() application
using as an argument for the Authenticate Application a variable defined on
every SIP or IAX account with setvar=mypasswd=1234 ( in sip.conf or iax.conf)
then you may put in your dialplan (extensions.conf) something like:
exten=>_X,1,Authenticate(${mypasswd}) ; where mypasswd is a variable
that exist on every sip account definition
exten=>_X,2,
http://www.voip-info.org/wiki-Asterisk+Database
http://www.voip-info.org/wiki-Asterisk+cmd+authenticate
Note this are just ideas, you must test it.
You may use Show application Authenticate and Show function DB_EXISTS within
Asterisk CLI to get a deeper understanding of all this functions and
applications
On 1/13/07, William Piper <[EMAIL PROTECTED]> wrote:
Try www.asterisk2billing.org
On 1/11/07, Pablo Bullian <[EMAIL PROTECTED]> wrote:
>
> Hi,
> I have an issue with the authentication for the outgoing calls.
>
> What I want is to give every user a different password, that they must
> enter everytime they make an outgoing call.
>
> What are my possibilities? and can u show me an example please?
>
> Thanks a lot.
>
> --
> 'May the source be with you'
>
> Pablo E. Bullian
> Network Administrator
> ___
> --Bandwidth and Colocation provided by Easynews.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
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Re: [asterisk-users] authentication issue!
Try www.asterisk2billing.org On 1/11/07, Pablo Bullian <[EMAIL PROTECTED]> wrote: Hi, I have an issue with the authentication for the outgoing calls. What I want is to give every user a different password, that they must enter everytime they make an outgoing call. What are my possibilities? and can u show me an example please? Thanks a lot. -- 'May the source be with you' Pablo E. Bullian Network Administrator ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] authentication issue!
Hi, I have an issue with the authentication for the outgoing calls. What I want is to give every user a different password, that they must enter everytime they make an outgoing call. What are my possibilities? and can u show me an example please? Thanks a lot. -- 'May the source be with you' Pablo E. Bullian Network Administrator ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
