If I recall 1.4.2 has a deadlock problem on their SIP channels -- you should upgrade to at least 1.4.5, which is when this was resolved. The problem was present in 1.2 and 1.4 -- 1.4.5 earliest that I believe does not have this issue.
Anthony On 8/17/07, Rizwan Hisham <[EMAIL PROTECTED]> wrote: > > Thanx for ur reply. > Im running * 1.4.2. i dont think there is any problem in asterisk because > only one user is having this problem. > > User is using Aastra 480i Cordless phone > Here is the sip config for that user. Im using call-limit=2 for every user > > [saadfarr] > username=saadfarr > type=friend > secret=123 > qualify=no > nat=yes > insecure=port,invite > call-limit=2 > host=dynamic > dtmfmode=rfc2833 > context=local > canreinvite=yes > accountcode=1:0:saadfarr > amaflags=default > > sip show channels give me the following: > Peer User/ANR Call ID Seq (Tx/Rx) Form Hold Last > Message > 68.144.208.153 (None) 1aed3833095 00101/00102 unkn No Rx: > OPTIONS > 85.234.144.137 (None) 256a1914164 00101/01429 unkn No Rx: > REGISTER > 216.143.130.70 1812923551 70f46ee82be 00102/00000 unkn No Tx: > ACK > 74.96.225.223 saadfarr 0c666dc33b3 00102/00000 unkn No > Init: INVITE > 74.96.225.223 saadfarr 522a18fd48c 00102/00000 unkn No > Init: INVITE > 66.131.246.220 foahand2 4e8cfe9c416 00102/00000 unkn No > Init: INVITE > 124.29.216.185 1212933903 443fdaeb50c 00102/00000 unkn No > Init: INVITE > 7 active SIP channels > > How do i know which one is dead/zombie channel. I can see 2 channels for > user saadfarr. i tried to use soft hangup but it requires channel > name.......how do i know the channel name if its a zombie channel. > > > > On 8/17/07, Anthony Cennami <[EMAIL PROTECTED]> wrote: > > > > Have you looked in show channels/core show channels to see if they have > > any dead/zombie channels, which you can remove with soft-hangup? > > > > What version of * are you running? > > > > What kind of phones? > > > > What config options are you using in SIP (or other tech) to limit the > > calls? > > > > > > On 8/17/07, Rizwan Hisham < [EMAIL PROTECTED]> wrote: > > > > > Hi all, > > > Some of my asterisk users have used their maximum call limit for > > > incoming calls (peers). There incoming call limit should automatically > > > reset > > > to zero after hangup but its not happening and they no longer can recieve > > > any calls as their allowed limit is already full. So is there any way to > > > reset the call limit on peers by commands or do i have to restart my > > > asterisk server? > > > > > > -- > > > Best Regards > > > Rizwan Hisham > > > Software Engineer > > > Axvoice Inc. > > > www.axvoice.com > > > _______________________________________________ > > > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > > > > > asterisk-users mailing list > > > To UNSUBSCRIBE or update options visit: > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > > > > > -- > > Anthony Cennami > > _______________________________________________ > > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > -- > Best Regards > Rizwan Hisham > Software Engineer > Axvoice Inc. > www.axvoice.com > > _______________________________________________ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- Anthony Cennami
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