Re: [asterisk-users] Digital accoustics trying to register to asterisk 1.4.43

2012-12-29 Thread Pete Mundy
On 14/12/2012, at 9:51 AM, Jerry Geis ge...@pagestation.com wrote:

 I did notice one more thing:
 chan_sip.c:17045 handle_request_register: Registration from 
 '5001sip:5001%4010.239.46.200@10.239.46.200' failed for '137.52.88.195' - 
 No matching peer found
 
 Why is there no matching peer I have it defined. I shows in my sip show 
 peers?

I wonder if in fact you have entered into the phone's web GUI the username 
5001@10.239.46.200 when you should have just entered 5001 (with the server 
name being defined elsewhere in the config, eg as the 'domain' value or the 
'proxy' value).

It looks to me as if the phone has encoded the string '5001@10.239.46.200' into 
the username '5001%4010.239.46.200' and then tried to connect to the server 
10.239.46.200 as that user (when in fact you actually want it to simply connect 
as '5001').

Worth trying? Could be a quick fix...

Pete



smime.p7s
Description: S/MIME cryptographic signature
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Digital accoustics trying to register to asterisk 1.4.43

2012-12-13 Thread Jerry Geis
I am trying to get a digital accoustics talkmaster to register to 
asterisk 1.4.43

I am getting the 401 unauthorized.

I have
host=dynamic
I have verified the passwords match

What else is there?

I dont see any further clues in sip set debug.
all it says is using request as basis request


What do I try?


--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
  http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [asterisk-users] Digital accoustics trying to register to asterisk 1.4.43

2012-12-13 Thread Danny Nicholas
Please post the sip.conf entry with any confidential data xxx'ed out.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis
Sent: Thursday, December 13, 2012 2:09 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Digital accoustics trying to register to asterisk
1.4.43

I am trying to get a digital accoustics talkmaster to register to asterisk
1.4.43 I am getting the 401 unauthorized.

I have
host=dynamic
I have verified the passwords match

What else is there?

I dont see any further clues in sip set debug.
all it says is using request as basis request


What do I try?


--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to
Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [asterisk-users] Digital accoustics trying to register to asterisk 1.4.43

2012-12-13 Thread Jerry Geis

[5001]
type=friend
username=5001
secret=XXX
dtmfmode=rfc2833
disallow=all
allow=ulaw
allow=alaw
context=incoming
host=dynamic
canreinvite=no
qualify=no
trustrpid=yes
sendrpid=no
nat=no



I did notice one more thing:
chan_sip.c:17045 handle_request_register: Registration from 
'5001sip:5001%4010.239.46.200@10.239.46.200' failed for '137.52.88.195' - 
No matching peer found

Why is there no matching peer I have it defined. I shows in my sip show peers?

jerry



--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
  http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [asterisk-users] Digital accoustics trying to register to asterisk 1.4.43

2012-12-13 Thread Danny Nicholas
The two things I would try are changing type from friend to peer and
sendrpid from no to yes.  The no matching peer usually means the device
username isn't matching the sip.conf username=.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis
Sent: Thursday, December 13, 2012 2:51 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Digital accoustics trying to register to
asterisk 1.4.43

[5001]
type=friend
username=5001
secret=XXX
dtmfmode=rfc2833
disallow=all
allow=ulaw
allow=alaw
context=incoming
host=dynamic
canreinvite=no
qualify=no
trustrpid=yes
sendrpid=no
nat=no



I did notice one more thing:
chan_sip.c:17045 handle_request_register: Registration from
'5001sip:5001%4010.239.46.200@10.239.46.200' failed for '137.52.88.195'
- No matching peer found

Why is there no matching peer I have it defined. I shows in my sip show
peers?

jerry



--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to
Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [asterisk-users] Digital accoustics trying to register to asterisk 1.4.43

2012-12-13 Thread Jerry Geis

The two things I would try are changing type from friend to peer and
sendrpid from no to yes.  The no matching peer usually means the device
username isn't matching the sip.conf username=.

I have tried both friend and peer. I changed the sendrpid to yes
and made no difference either. Still get 401 Unauthorized.

Jerry


--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Digital accoustics trying to register to asterisk 1.4.43

2012-12-13 Thread Danny Nicholas
This animal might be like the OBI110 box where you set it up in users.conf
instead of sip.conf.

Something like this:

[5001]

transfer=yes

call-limit=5

registersip=no

host = 1.2.3.4

context=default

hasvoicemail=no

dtmfmode=inband

threewaycalling=no

hasdirectory=no

callwaiting=no

hasmanager=no

managerread = system,call,log,verbose,command,agent,user,config

managerwrite = system,call,log,verbose,command,agent,user,config

hasagent = no

hassip=yes

hasiax=no

secret=x

nat=no

canreinvite=no

dtmfmode=rfc2833

insecure=port,invite

pickupgroup=1

callgroup=1

disallow = all

allow = ulaw,gsm

 

You still do sip reload to get it connected.

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis
Sent: Thursday, December 13, 2012 3:01 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Digital accoustics trying to register to
asterisk 1.4.43

 

The two things I would try are changing type from friend to peer and
sendrpid from no to yes.  The no matching peer usually means the device
username isn't matching the sip.conf username=.

I have tried both friend and peer. I changed the sendrpid to yes 
and made no difference either. Still get 401 Unauthorized.

Jerry



--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Digital accoustics trying to register to asterisk 1.4.43

2012-12-13 Thread Jerry Geis

This animal might be like the OBI110 box where you set it up in users.conf
instead of sip.conf.

Something like this:

[5001]

transfer=yes

call-limit=5

registersip=no

host = 1.2.3.4

context=default

hasvoicemail=no

dtmfmode=inband

threewaycalling=no

hasdirectory=no

callwaiting=no

hasmanager=no

managerread = system,call,log,verbose,command,agent,user,config

managerwrite = system,call,log,verbose,command,agent,user,config

hasagent = no

hassip=yes

hasiax=no

secret=x

nat=no

canreinvite=no

dtmfmode=rfc2833

insecure=port,invite

pickupgroup=1

callgroup=1

disallow = all

allow = ulaw,gsm



You still do sip reload to get it connected.


That worked - it registered.

Why would it not register the other way?

Jerry

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Digital accoustics trying to register to asterisk 1.4.43

2012-12-13 Thread Danny Nicholas
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis
Sent: Thursday, December 13, 2012 3:34 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Digital accoustics trying to register to
asterisk 1.4.43

 

This animal might be like the OBI110 box where you set it up in users.conf
instead of sip.conf.
 
Something like this:
 
[5001]
 
transfer=yes
 
call-limit=5
 
registersip=no
 
host = 1.2.3.4
 
context=default
 
hasvoicemail=no
 
dtmfmode=inband
 
threewaycalling=no
 
hasdirectory=no
 
callwaiting=no
 
hasmanager=no
 
managerread = system,call,log,verbose,command,agent,user,config
 
managerwrite = system,call,log,verbose,command,agent,user,config
 
hasagent = no
 
hassip=yes
 
hasiax=no
 
secret=x
 
nat=no
 
canreinvite=no
 
dtmfmode=rfc2833
 
insecure=port,invite
 
pickupgroup=1
 
callgroup=1
 
disallow = all
 
allow = ulaw,gsm
 
 
 
You still do sip reload to get it connected.

 

That worked - it registered.

Why would it not register the other way?

Jerry

n  It's supposed to work both ways.  It depends on how you have it set up on
the remote side.  It's been two years since I went through the process so it
isn't fresh on my brain.

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users