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hawat.thufir at gmail.com
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 Posted: Fri Feb 20, 2015 6:15 am Post subject: [asterisk-users] sipsak 200 for a user, but 404 for a differ |
 |
|
What's the difference between user "123" and "devries"? Based on the
output here, they seem the same..?
tleilax*CLI>
tleilax*CLI> sip show users
Username Secret Accountcode
Def.Context ACL Forcerport
201 password 201
default No Yes
123 password 123
default No Yes
devries password devries
default No Yes
babytel hjkgk58757
default No Yes
gs102 X58sKpZCcDfcGT0 gs102
default No Yes
tleilax*CLI>
tleilax*CLI> sip show user 123
* Name : 123
Secret : <Set>
MD5Secret : <Not set>
Context : default
Language : en
Accountcode : 123
AMA flags : Unknown
Netborder CPD: No
Transfer mode: open
MaxCallBR : 384 kbps
CallingPres : Presentation Allowed, Not Screened
Call limit : 0
Callgroup :
Pickupgroup :
Callerid : "123" <123>
ACL : No
Sess-Timers : Accept
Sess-Refresh : uas
Sess-Expires : 1800 secs
Sess-Min-SE : 90 secs
RTP Engine : asterisk
Codec Order : (ulaw:20,gsm:20)
Auto-Framing: No
tleilax*CLI>
tleilax*CLI> sip show user devries
* Name : devries
Secret : <Set>
MD5Secret : <Not set>
Context : default
Language : en
Accountcode : devries
AMA flags : Unknown
Netborder CPD: No
Transfer mode: open
MaxCallBR : 384 kbps
CallingPres : Presentation Allowed, Not Screened
Call limit : 0
Callgroup :
Pickupgroup :
Callerid : "devries" <999>
ACL : No
Sess-Timers : Accept
Sess-Refresh : uas
Sess-Expires : 1800 secs
Sess-Min-SE : 90 secs
RTP Engine : asterisk
Codec Order : (ulaw:20,gsm:20)
Auto-Framing: No
tleilax*CLI>
tleilax*CLI> exit
tleilax:~ #
tleilax:~ # exit
logout
Connection to tleilax closed.
thufir@doge:~$
thufir@doge:~$ sudo sipsak -vv -s sip:123@tleilax
[sudo] password for thufir:
No SRV record: _sip._tcp.tleilax
No SRV record: _sip._udp.tleilax
using A record: tleilax
message received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
127.0.1.1:55238;branch=z9hG4bK.3e59b63f;alias;received=192.168.1.3;rport=55238
From: sip:sipsak@127.0.1.1:55238;tag=1e6fe4eb
To: sip:123@tleilax;tag=as7dc4727d
Call-ID: 510649579@127.0.1.1
CSeq: 1 OPTIONS
Server: Asterisk PBX 1.8.32.1-vici
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:192.168.1.2:5060>
Accept: application/sdp
Content-Length: 0
** reply received after 0.627 ms **
SIP/2.0 200 OK
final received
thufir@doge:~$
thufir@doge:~$ sudo sipsak -vv -s sip:devries@tleilax
No SRV record: _sip._tcp.tleilax
No SRV record: _sip._udp.tleilax
using A record: tleilax
message received:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP
127.0.1.1:54969;branch=z9hG4bK.38ee5c41;alias;received=192.168.1.3;rport=54969
From: sip:sipsak@127.0.1.1:54969;tag=6e148be1
To: sip:devries@tleilax;tag=as2b617a9b
Call-ID: 1846840289@127.0.1.1
CSeq: 1 OPTIONS
Server: Asterisk PBX 1.8.32.1-vici
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces, timer
Accept: application/sdp
Content-Length: 0
** reply received after 0.648 ms **
SIP/2.0 404 Not Found
final received
thufir@doge:~$
thanks,
Thufir
--
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andres at telesip.net
Guest
|
| Posted: Fri Feb 20, 2015 8:46 am Post subject: [asterisk-users] sipsak 200 for a user, but 404 for a differ |
|
|
On 2/20/15 6:15 AM, thufir wrote:
| Quote: | What's the difference between user "123" and "devries"? Based on the
output here, they seem the same..?
tleilax*CLI>
tleilax*CLI> sip show users
Username Secret Accountcode
Def.Context ACL Forcerport
201 password 201
default No Yes
123 password 123
default No Yes
devries password devries
default No Yes
babytel hjkgk58757
default No Yes
gs102 X58sKpZCcDfcGT0 gs102
default No Yes
tleilax*CLI>
tleilax*CLI> sip show user 123
* Name : 123
Secret : <Set>
MD5Secret : <Not set>
Context : default
Language : en
Accountcode : 123
AMA flags : Unknown
Netborder CPD: No
Transfer mode: open
MaxCallBR : 384 kbps
CallingPres : Presentation Allowed, Not Screened
Call limit : 0
Callgroup :
Pickupgroup :
Callerid : "123" <123>
ACL : No
Sess-Timers : Accept
Sess-Refresh : uas
Sess-Expires : 1800 secs
Sess-Min-SE : 90 secs
RTP Engine : asterisk
Codec Order : (ulaw:20,gsm:20)
Auto-Framing: No
tleilax*CLI>
tleilax*CLI> sip show user devries
* Name : devries
Secret : <Set>
MD5Secret : <Not set>
Context : default
Language : en
Accountcode : devries
AMA flags : Unknown
Netborder CPD: No
Transfer mode: open
MaxCallBR : 384 kbps
CallingPres : Presentation Allowed, Not Screened
Call limit : 0
Callgroup :
Pickupgroup :
Callerid : "devries" <999>
ACL : No
Sess-Timers : Accept
Sess-Refresh : uas
Sess-Expires : 1800 secs
Sess-Min-SE : 90 secs
RTP Engine : asterisk
Codec Order : (ulaw:20,gsm:20)
Auto-Framing: No
tleilax*CLI>
tleilax*CLI> exit
tleilax:~ #
tleilax:~ # exit
logout
Connection to tleilax closed.
thufir@doge:~$
thufir@doge:~$ sudo sipsak -vv -s sip:123@tleilax
[sudo] password for thufir:
No SRV record: _sip._tcp.tleilax
No SRV record: _sip._udp.tleilax
using A record: tleilax
message received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
127.0.1.1:55238;branch=z9hG4bK.3e59b63f;alias;received=192.168.1.3;rport=55238
From: sip:sipsak@127.0.1.1:55238;tag=1e6fe4eb
To: sip:123@tleilax;tag=as7dc4727d
Call-ID: 510649579@127.0.1.1
CSeq: 1 OPTIONS
Server: Asterisk PBX 1.8.32.1-vici
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:192.168.1.2:5060>
Accept: application/sdp
Content-Length: 0
** reply received after 0.627 ms **
SIP/2.0 200 OK
final received
thufir@doge:~$
thufir@doge:~$ sudo sipsak -vv -s sip:devries@tleilax
No SRV record: _sip._tcp.tleilax
No SRV record: _sip._udp.tleilax
using A record: tleilax
message received:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP
127.0.1.1:54969;branch=z9hG4bK.38ee5c41;alias;received=192.168.1.3;rport=54969
From: sip:sipsak@127.0.1.1:54969;tag=6e148be1
To: sip:devries@tleilax;tag=as2b617a9b
Call-ID: 1846840289@127.0.1.1
CSeq: 1 OPTIONS
Server: Asterisk PBX 1.8.32.1-vici
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces, timer
Accept: application/sdp
Content-Length: 0
** reply received after 0.648 ms **
SIP/2.0 404 Not Found
final received
thufir@doge:~$
|
A "sip set debug on" will give you more info on why you are getting the
404. It probably has to do something with your context/dialplan.
--
Technical Support
http://www.cellroute.net
--
_____________________________________________________________________
-- 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:
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hawat.thufir at gmail.com
Guest
|
| Posted: Fri Feb 20, 2015 2:29 pm Post subject: [asterisk-users] sipsak 200 for a user, but 404 for a differ |
|
|
On Fri, 20 Feb 2015 08:46:13 -0500, Andres wrote:
| Quote: | A "sip set debug on" will give you more info on why you are getting the
404. It probably has to do something with your context/dialplan.
|
on tleilax:
tleilax*CLI>
tleilax*CLI> sip set debug on
SIP Debugging enabled
tleilax*CLI>
on doge:
thufir@doge:~$
thufir@doge:~$ sudo sipsak -vv -s sip:devries@tleilax -m "hi"
No SRV record: _sip._tcp.tleilax
No SRV record: _sip._udp.tleilax
using A record: tleilax
Max-Forwards set to 0
message received:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP
127.0.1.1:56377;branch=z9hG4bK.0edaada3;alias;received=192.168.1.3;rport=56377
From: sip:sipsak@127.0.1.1:56377;tag=6b540010
To: sip:devries@tleilax;tag=as02b0fdd6
Call-ID: 1800667152@127.0.1.1
CSeq: 1 OPTIONS
Server: Asterisk PBX 1.8.32.1-vici
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces, timer
Accept: application/sdp
Content-Length: 0
** reply received after 0.844 ms **
SIP/2.0 404 Not Found
final received
thufir@doge:~$
However, I'm sure you're right that it's the dialplan; I'm looking into
it.
thanks,
Thufir
--
_____________________________________________________________________
-- 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 |
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andres at telesip.net
Guest
|
| Posted: Fri Feb 20, 2015 3:07 pm Post subject: [asterisk-users] sipsak 200 for a user, but 404 for a differ |
|
|
On 2/20/15 2:29 PM, thufir wrote:
| Quote: | On Fri, 20 Feb 2015 08:46:13 -0500, Andres wrote:
| Quote: | A "sip set debug on" will give you more info on why you are getting the
404. It probably has to do something with your context/dialplan.
|
on tleilax:
tleilax*CLI>
tleilax*CLI> sip set debug on
SIP Debugging enabled
tleilax*CLI>
|
This is showing nothing so I don't think your test message even made it
here. I think it looped in the 'doge' server.
| Quote: |
on doge:
thufir@doge:~$
thufir@doge:~$ sudo sipsak -vv -s sip:devries@tleilax -m "hi"
No SRV record: _sip._tcp.tleilax
No SRV record: _sip._udp.tleilax
using A record: tleilax
Max-Forwards set to 0
message received:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP
127.0.1.1:56377;branch=z9hG4bK.0edaada3;alias;received=192.168.1.3;rport=56377
From: sip:sipsak@127.0.1.1:56377;tag=6b540010
To: sip:devries@tleilax;tag=as02b0fdd6
Call-ID: 1800667152@127.0.1.1
CSeq: 1 OPTIONS
Server: Asterisk PBX 1.8.32.1-vici
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces, timer
Accept: application/sdp
Content-Length: 0
** reply received after 0.844 ms **
SIP/2.0 404 Not Found
final received
thufir@doge:~$
However, I'm sure you're right that it's the dialplan; I'm looking into
it.
thanks,
Thufir
|
--
Technical Support
http://www.cellroute.net
--
_____________________________________________________________________
-- 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 |
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| Back to top |
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hawat.thufir at gmail.com
Guest
|
| Posted: Fri Feb 20, 2015 3:21 pm Post subject: [asterisk-users] sipsak 200 for a user, but 404 for a differ |
|
|
On Fri, 20 Feb 2015 15:07:35 -0500, Andres wrote:
| Quote: | This is showing nothing so I don't think your test message even made it
here. I think it looped in the 'doge' server.
|
I was wondering the same thing 
in tleilax, I looked in /var/log/asterisk/messages and see:
[Feb 20 15:13:19] VERBOSE[3661] chan_sip.c: [Feb 20 15:13:19]
<--- SIP read from UDP:192.168.1.3:38154 --->
OPTIONS sip:345@tleilax SIP/2.0
Via: SIP/2.0/UDP 127.0.1.1:38154;branch=z9hG4bK.77fd156e;rport;alias
From: sip:sipsak@127.0.1.1:38154;tag=4653e713
To: sip:345@tleilax
Call-ID: 1179903763@127.0.1.1
CSeq: 1 OPTIONS
Contact: sip:sipsak@127.0.1.1:38154
Content-Length: 0
Max-Forwards: 0
User-Agent: sipsak 0.9.6
Accept: text/plain
it seems to work, in that I get 200 OK, with success reflected,
apparently, in the log, provided that its numerical. I just changed it
from "piter" to "345" and get success (well, at this at least). This
probably has something to do with my dialplan..
Is the message, "hi", logged anywhere?
-Thufir
--
_____________________________________________________________________
-- 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 |
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andres at telesip.net
Guest
|
| Posted: Fri Feb 20, 2015 5:12 pm Post subject: [asterisk-users] sipsak 200 for a user, but 404 for a differ |
|
|
On 2/20/15 3:20 PM, thufir wrote:
| Quote: | On Fri, 20 Feb 2015 15:07:35 -0500, Andres wrote:
| Quote: | This is showing nothing so I don't think your test message even made it
here. I think it looped in the 'doge' server.
|
I was wondering the same thing
in tleilax, I looked in /var/log/asterisk/messages and see:
[Feb 20 15:13:19] VERBOSE[3661] chan_sip.c: [Feb 20 15:13:19]
<--- SIP read from UDP:192.168.1.3:38154 --->
OPTIONS sip:345@tleilax SIP/2.0
Via: SIP/2.0/UDP 127.0.1.1:38154;branch=z9hG4bK.77fd156e;rport;alias
From: sip:sipsak@127.0.1.1:38154;tag=4653e713
To: sip:345@tleilax
Call-ID: 1179903763@127.0.1.1
CSeq: 1 OPTIONS
Contact: sip:sipsak@127.0.1.1:38154
Content-Length: 0
Max-Forwards: 0
User-Agent: sipsak 0.9.6
Accept: text/plain
it seems to work, in that I get 200 OK, with success reflected,
apparently, in the log, provided that its numerical. I just changed it
from "piter" to "345" and get success (well, at this at least). This
probably has something to do with my dialplan..
Is the message, "hi", logged anywhere?
|
I don't think so. But you should also see the SIP messages on the
console (sip set debug on) without having to look at the log file. Maybe
something in your logger.conf is messed up.
--
Technical Support
http://www.cellroute.net
--
_____________________________________________________________________
-- 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 |
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hawat.thufir at gmail.com
Guest
|
| Posted: Fri Feb 20, 2015 9:10 pm Post subject: [asterisk-users] sipsak 200 for a user, but 404 for a differ |
|
|
On Fri, 20 Feb 2015 17:11:53 -0500, Andres wrote:
| Quote: | I don't think so. But you should also see the SIP messages on the
console (sip set debug on) without having to look at the log file. Maybe
something in your logger.conf is messed up.
|
that worked
tleilax*CLI>
tleilax*CLI>
[Feb 20 21:06:19]
<--- SIP read from UDP:192.168.1.3:44226 --->
OPTIONS sip:345@tleilax SIP/2.0
Via: SIP/2.0/UDP 127.0.1.1:44226;branch=z9hG4bK.508a6d72;rport;alias
From: sip:sipsak@127.0.1.1:44226;tag=2a099edc
To: sip:345@tleilax
Call-ID: 705273564@127.0.1.1
CSeq: 1 OPTIONS
Contact: sip:sipsak@127.0.1.1:44226
Content-Length: 0
Max-Forwards: 0
User-Agent: sipsak 0.9.6
Accept: text/plain
<------------->
[Feb 20 21:06:19] --- (11 headers 0 lines) ---
[Feb 20 21:06:19] Looking for 345 in trunkinbound (domain tleilax)
[Feb 20 21:06:19]
<--- Transmitting (NAT) to 192.168.1.3:44226 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
127.0.1.1:44226;branch=z9hG4bK.508a6d72;alias;received=192.168.1.3;rport=44226
From: sip:sipsak@127.0.1.1:44226;tag=2a099edc
To: sip:345@tleilax;tag=as5d21da5c
Call-ID: 705273564@127.0.1.1
CSeq: 1 OPTIONS
Server: Asterisk PBX 1.8.29.0-vici
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:192.168.1.2:5060>
Accept: application/sdp
Content-Length: 0
<------------>
[Feb 20 21:06:19] Scheduling destruction of SIP dialog
'705273564@127.0.1.1' in 32000 ms (Method: OPTIONS)
[Feb 20 21:06:38] Really destroying SIP dialog '1876256264@127.0.1.1'
Method: OPTIONS
[Feb 20 21:06:51] Really destroying SIP dialog '705273564@127.0.1.1'
Method: OPTIONS
tleilax*CLI> exit
tleilax:~ #
I would've liked to see the "hi" message, but it's good to see that
result server side.
-Thufir
--
_____________________________________________________________________
-- 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 |
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