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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070817/9706efcf/attachment.htm
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 <rizwanhasham at gmail.com> 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070817/b2da16bf/attachment.htm
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 <acennami at gmail.com> 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 <rizwanhasham at gmail.com> 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070817/7f77ffd4/attachment.htm
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 <rizwanhasham at gmail.com> 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 <acennami at gmail.com> 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 < rizwanhasham at gmail.com> 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070817/fc4cd773/attachment.htm
At 06:37 AM 8/17/2007, you wrote:>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?It's not just you, it happens to my wife too. No rhyme or reason I can see, I just try to restart asterisk occasionally so it doesn't get that far. Ira