[asterisk-users] Call Limits

[Anthony Cennami]

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
> > > _______________________________________________
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> > >
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> > >
> >
> >
> >
> > --
> > Anthony Cennami
> > _______________________________________________
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> >
> > asterisk-users mailing list
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> >
>
>
>
> --
> 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
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-- 
Anthony Cennami
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