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[asterisk-users] NAME/USERNAME conflict


 
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kazabe at gmail.com
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PostPosted: Thu Jan 21, 2016 10:50 am    Post subject: [asterisk-users] NAME/USERNAME conflict Reply with quote

Hi.

we are experimenting a strange issue in our PBX.


By example: if we dial to the 100, the call is answered in 199.  We dont have any redirection for that, but the cli show the same issue when request show peers.  Aditionally, the user 100 use the ip address 192.168.11.100, and the cli show connected the user from 192.168.11.160 (that ip is assigned to the user 199)


PBX*CLI> sip show peers
Name/username             Host                                    Dyn Forcerport Comedia    ACL Port     Status      Description
100/199                   192.168.11.160                           D  Yes        Yes         A  5060     OK (30 ms)           





I check the sip 100 and (aparently) show all normal





PBX*CLI> sip show peer 100




  * Name       : 100
  Description  :
  Secret       : <Set>
  MD5Secret    : <Not set>
  Remote Secret: <Not set>
  Context      : MAIN
  Record On feature : automon
  Record Off feature : automon
  Subscr.Cont. : <Not set>
  Language     :
  Tonezone     : <Not set>
  AMA flags    : Unknown
  Transfer mode: open
  CallingPres  : Presentation Allowed, Not Screened
  Callgroup    : 1
  Pickupgroup  : 1
  Named Callgr :
  Nam. Pickupgr:
  MOH Suggest  :
  Mailbox      : 100@device
  VM Extension : *97
  LastMsgsSent : 0/0
  Call limit   : 2147483647
  Max forwards : 0
  Dynamic      : Yes
  Callerid     : "JDOE" <100>
  MaxCallBR    : 384 kbps
  Expire       : 2680
  Insecure     : no
  Force rport  : Yes
  Symmetric RTP: Yes
  ACL          : Yes
  DirectMedACL : No
  T.38 support : No
  T.38 EC mode : Unknown
  T.38 MaxDtgrm: 4294967295
  DirectMedia  : No
  PromiscRedir : No
  User=Phone   : No
  Video Support: No
  Text Support : No
  Ign SDP ver  : No
  Trust RPID   : Yes
  Send RPID    : No
  TrustIDOutbnd: Legacy
  Subscriptions: Yes
  Overlap dial : Yes
  DTMFmode     : rfc2833
  Timer T1     : 500
  Timer B      : 32000
  ToHost       :
  Addr->IP     : 192.168.11.160:5060
  Defaddr->IP  : (null)
  Prim.Transp. : UDP
  Allowed.Trsp : UDP
  Def. Username: 199
  SIP Options  : path replaces replace timer
  Codecs       : (ulaw)
  Codec Order  : (ulaw:20)
  Auto-Framing : No
  Status       : OK (28 ms)
  Useragent    : Grandstream GXP2000 1.2.5.3
  Reg. Contact : sip:101@192.168.11.160:5060;transport=udp
  Qualify Freq : 60000 ms
  Keepalive    : 0 ms
  Sess-Timers  : Accept
  Sess-Refresh : uas
  Sess-Expires : 1800 secs
  Min-Sess     : 90 secs
  RTP Engine   : asterisk
  Parkinglot   :
  Use Reason   : No
  Encryption   : No



What can cause that?  i delete both extensions and create again and the problem continue.  Adn others extensions are showing the same issue (call to another extension and answer at 199).


thanks in advance
Back to top
ethy.brito at inexo.co...
Guest





PostPosted: Thu Jan 21, 2016 10:59 am    Post subject: [asterisk-users] NAME/USERNAME conflict Reply with quote

On Thu, 21 Jan 2016 10:49:21 -0500
kazabe <kazabe@gmail.com> wrote:

Quote:
Hi.

we are experimenting a strange issue in our PBX.

By example: if we dial to the 100, the call is answered in 199. We dont
have any redirection for that, but the cli show the same issue when request
show peers. Aditionally, the user 100 use the ip address 192.168.11.100,
and the cli show connected the user from 192.168.11.160 (that ip is
assigned to the user 199)

PBX*CLI> sip show peers
Name/username Host Dyn
Forcerport Comedia ACL Port Status Description
100/199 192.168.11.160 D Yes
Yes A 5060 OK (30 ms)


I check the sip 100 and (aparently) show all normal


PBX*CLI> sip show peer 100


* Name : 100
Description :
Secret : <Set>
MD5Secret : <Not set>
Remote Secret: <Not set>
Context : MAIN
Record On feature : automon
Record Off feature : automon
Subscr.Cont. : <Not set>
Language :
Tonezone : <Not set>
AMA flags : Unknown
Transfer mode: open
CallingPres : Presentation Allowed, Not Screened
Callgroup : 1
Pickupgroup : 1
Named Callgr :
Nam. Pickupgr:
MOH Suggest :
Mailbox : 100@device
VM Extension : *97
LastMsgsSent : 0/0
Call limit : 2147483647
Max forwards : 0
Dynamic : Yes
Callerid : "JDOE" <100>
MaxCallBR : 384 kbps
Expire : 2680
Insecure : no
Force rport : Yes
Symmetric RTP: Yes
ACL : Yes
DirectMedACL : No
T.38 support : No
T.38 EC mode : Unknown
T.38 MaxDtgrm: 4294967295
DirectMedia : No
PromiscRedir : No
User=Phone : No
Video Support: No
Text Support : No
Ign SDP ver : No
Trust RPID : Yes
Send RPID : No
TrustIDOutbnd: Legacy
Subscriptions: Yes
Overlap dial : Yes
DTMFmode : rfc2833
Timer T1 : 500
Timer B : 32000
ToHost :
Addr->IP : 192.168.11.160:5060
Defaddr->IP : (null)
Prim.Transp. : UDP
Allowed.Trsp : UDP
Def. Username: 199
SIP Options : path replaces replace timer
Codecs : (ulaw)
Codec Order : (ulaw:20)
Auto-Framing : No
Status : OK (28 ms)
Useragent : Grandstream GXP2000 1.2.5.3
Reg. Contact : sip:101@192.168.11.160:5060;transport=udp
Qualify Freq : 60000 ms
Keepalive : 0 ms
Sess-Timers : Accept
Sess-Refresh : uas
Sess-Expires : 1800 secs
Min-Sess : 90 secs
RTP Engine : asterisk
Parkinglot :
Use Reason : No
Encryption : No

What can cause that? i delete both extensions and create again and the
problem continue. Adn others extensions are showing the same issue (call
to another extension and answer at 199).

thanks in advance

Hi

Check if the extensions 100 and 109 aren't using the same username to register themselves.

Cheers

Ethy




--

Ethy H. Brito /"\
InterNexo Ltda. \ / CAMPANHA DA FITA ASCII - CONTRA MAIL HTML
+55 (12) 3797-6860 X ASCII RIBBON CAMPAIGN - AGAINST HTML MAIL
S.J.Campos - Brasil / \

PGP key: http://www.inexo.com.br/~ethy/0xC3F222A0.asc

--
_____________________________________________________________________
-- 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
Back to top
kazabe at gmail.com
Guest





PostPosted: Thu Jan 21, 2016 7:37 pm    Post subject: [asterisk-users] NAME/USERNAME conflict Reply with quote

hi.
thanks for the recommendation.  i discard that (and delete and create again the ext with random generated password), and the issue continue. El ene 21, 2016 10:59 AM, "Ethy H. Brito" <ethy.brito@inexo.com.br (ethy.brito@inexo.com.br)> escribió:
Quote:
On Thu, 21 Jan 2016 10:49:21 -0500
kazabe <kazabe@gmail.com (kazabe@gmail.com)> wrote:

Quote:
Hi.

we are experimenting a strange issue in our PBX.

By example: if we dial to the 100, the call is answered in 199.  We dont
have any redirection for that, but the cli show the same issue when request
show peers.  Aditionally, the user 100 use the ip address 192.168.11.100,
and the cli show connected the user from 192.168.11.160 (that ip is
assigned to the user 199)

PBX*CLI> sip show peers
Name/username             Host                                    Dyn
Forcerport Comedia    ACL Port     Status      Description
100/199                   192.168.11.160                           D  Yes
      Yes         A  5060     OK (30 ms)


I check the sip 100 and (aparently) show all normal


PBX*CLI> sip show peer 100


   * Name       : 100
   Description  :
   Secret       : <Set>
   MD5Secret    : <Not set>
   Remote Secret: <Not set>
   Context      : MAIN
   Record On feature : automon
   Record Off feature : automon
   Subscr.Cont. : <Not set>
   Language     :
   Tonezone     : <Not set>
   AMA flags    : Unknown
   Transfer mode: open
   CallingPres  : Presentation Allowed, Not Screened
   Callgroup    : 1
   Pickupgroup  : 1
   Named Callgr :
   Nam. Pickupgr:
   MOH Suggest  :
   Mailbox      : 100@device
   VM Extension : *97
   LastMsgsSent : 0/0
   Call limit   : [url=tel:2147483647]2147483647[/url]
   Max forwards : 0
   Dynamic      : Yes
   Callerid     : "JDOE" <100>
   MaxCallBR    : 384 kbps
   Expire       : 2680
   Insecure     : no
   Force rport  : Yes
   Symmetric RTP: Yes
   ACL          : Yes
   DirectMedACL : No
   T.38 support : No
   T.38 EC mode : Unknown
   T.38 MaxDtgrm: 4294967295
   DirectMedia  : No
   PromiscRedir : No
   User=Phone   : No
   Video Support: No
   Text Support : No
   Ign SDP ver  : No
   Trust RPID   : Yes
   Send RPID    : No
   TrustIDOutbnd: Legacy
   Subscriptions: Yes
   Overlap dial : Yes
   DTMFmode     : rfc2833
   Timer T1     : 500
   Timer B      : 32000
   ToHost       :
   Addr->IP     : 192.168.11.160:5060
   Defaddr->IP  : (null)
   Prim.Transp. : UDP
   Allowed.Trsp : UDP
   Def. Username: 199
   SIP Options  : path replaces replace timer
   Codecs       : (ulaw)
   Codec Order  : (ulaw:20)
   Auto-Framing : No
   Status       : OK (28 ms)
   Useragent    : Grandstream GXP2000 1.2.5.3
   Reg. Contact : sip:101@192.168.11.160:5060;transport=udp
   Qualify Freq : 60000 ms
   Keepalive    : 0 ms
   Sess-Timers  : Accept
   Sess-Refresh : uas
   Sess-Expires : 1800 secs
   Min-Sess     : 90 secs
   RTP Engine   : asterisk
   Parkinglot   :
   Use Reason   : No
   Encryption   : No

What can cause that?  i delete both extensions and create again and the
problem continue.  Adn others extensions are showing the same issue (call
to another extension and answer at 199).

thanks in advance

Hi

Check if the extensions 100 and 109 aren't using the same username to register themselves.

Cheers

Ethy




--

Ethy H. Brito         /"\
InterNexo Ltda.       \ /  CAMPANHA DA FITA ASCII - CONTRA MAIL HTML
[url=tel:%2B55%20%2812%29%203797-6860]+55 (12) 3797-6860[/url]     X   ASCII RIBBON CAMPAIGN - AGAINST HTML MAIL
S.J.Campos - Brasil   / \

PGP key: http://www.inexo.com.br/~ethy/0xC3F222A0.asc

--
_____________________________________________________________________
-- 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
Back to top
ethy.brito at inexo.co...
Guest





PostPosted: Thu Jan 21, 2016 9:19 pm    Post subject: [asterisk-users] NAME/USERNAME conflict Reply with quote

On Thu, 21 Jan 2016 19:36:42 -0500
kazabe <kazabe@gmail.com> wrote:

Quote:
hi.

thanks for the recommendation. i discard that (and delete and create
again the ext with random generated password), and the issue continue.

Sorry. I meant username/password you set on both terminals, not in asterisk's 'conf' files.

Also check if the IP addresses on the terminals are really those you wrote here and
the ether addresses of both if they are on the same ethernet segment
that asterisk is with
arp -an | grep '192.168.11.\(100\|160\)'

Can you "ping" them from asterisk?

What are the outputs of
sip show peer 100
and
sip show peer 109
?

Note that your 'sip show peers' shows only one registered terminal. It
means, in your case, that asterisk may be getting the same registration
credentials from two different sources (IPs).

Cheers

Ethy


Quote:
El ene 21, 2016 10:59 AM, "Ethy H. Brito" <ethy.brito@inexo.com.br>
escribió:

Quote:
On Thu, 21 Jan 2016 10:49:21 -0500
kazabe <kazabe@gmail.com> wrote:

Quote:
Hi.

we are experimenting a strange issue in our PBX.

By example: if we dial to the 100, the call is answered in 199. We
dont have any redirection for that, but the cli show the same issue
when
request
Quote:
show peers. Aditionally, the user 100 use the ip address
192.168.11.100, and the cli show connected the user from
192.168.11.160 (that ip is assigned to the user 199)

PBX*CLI> sip show peers
Name/username Host Dyn
Forcerport Comedia ACL Port Status Description
100/199 192.168.11.160
D Yes Yes A 5060 OK (30 ms)


I check the sip 100 and (aparently) show all normal


PBX*CLI> sip show peer 100


* Name : 100
Description :
Secret : <Set>
MD5Secret : <Not set>
Remote Secret: <Not set>
Context : MAIN
Record On feature : automon
Record Off feature : automon
Subscr.Cont. : <Not set>
Language :
Tonezone : <Not set>
AMA flags : Unknown
Transfer mode: open
CallingPres : Presentation Allowed, Not Screened
Callgroup : 1
Pickupgroup : 1
Named Callgr :
Nam. Pickupgr:
MOH Suggest :
Mailbox : 100@device
VM Extension : *97
LastMsgsSent : 0/0
Call limit : 2147483647
Max forwards : 0
Dynamic : Yes
Callerid : "JDOE" <100>
MaxCallBR : 384 kbps
Expire : 2680
Insecure : no
Force rport : Yes
Symmetric RTP: Yes
ACL : Yes
DirectMedACL : No
T.38 support : No
T.38 EC mode : Unknown
T.38 MaxDtgrm: 4294967295
DirectMedia : No
PromiscRedir : No
User=Phone : No
Video Support: No
Text Support : No
Ign SDP ver : No
Trust RPID : Yes
Send RPID : No
TrustIDOutbnd: Legacy
Subscriptions: Yes
Overlap dial : Yes
DTMFmode : rfc2833
Timer T1 : 500
Timer B : 32000
ToHost :
Addr->IP : 192.168.11.160:5060
Defaddr->IP : (null)
Prim.Transp. : UDP
Allowed.Trsp : UDP
Def. Username: 199
SIP Options : path replaces replace timer
Codecs : (ulaw)
Codec Order : (ulaw:20)
Auto-Framing : No
Status : OK (28 ms)
Useragent : Grandstream GXP2000 1.2.5.3
Reg. Contact : sip:101@192.168.11.160:5060;transport=udp
Qualify Freq : 60000 ms
Keepalive : 0 ms
Sess-Timers : Accept
Sess-Refresh : uas
Sess-Expires : 1800 secs
Min-Sess : 90 secs
RTP Engine : asterisk
Parkinglot :
Use Reason : No
Encryption : No

What can cause that? i delete both extensions and create again and
the problem continue. Adn others extensions are showing the same
issue (call to another extension and answer at 199).

thanks in advance

Hi

Check if the extensions 100 and 109 aren't using the same username to
register themselves.

Cheers

Ethy




--

Ethy H. Brito /"\
InterNexo Ltda. \ / CAMPANHA DA FITA ASCII - CONTRA MAIL HTML
+55 (12) 3797-6860 X ASCII RIBBON CAMPAIGN - AGAINST HTML MAIL
S.J.Campos - Brasil / \

PGP key: http://www.inexo.com.br/~ethy/0xC3F222A0.asc

--
_____________________________________________________________________
-- 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
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