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 at 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 at 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
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
<http://lists.digium.com/pipermail/asterisk-users/attachments/20160121/441674f7/attachment.html>
On Thu, 21 Jan 2016 10:49:21 -0500 kazabe <kazabe at gmail.com> wrote:> 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 at 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 at 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 advanceHi 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
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 at inexo.com.br> escribi?:> On Thu, 21 Jan 2016 10:49:21 -0500 > kazabe <kazabe at gmail.com> wrote: > > > 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 at 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 at 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 >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160121/13a50bd0/attachment.html>