My cdrs are missing accountcodes for incoming calls from other asterisk
servers..
I've seen a few people mentioning this on the list and the solution
seems to be setting up a dialplan for incoming calls from a particular
sip peer.. in my opinion this does not scale well at all and I am
looking for a solution to correct this problem.
example sip peer:
[asterisk_gw]
type=friend
accountcode=asteriskgw
host=x.x.x.x
restrictcid=no
fromdomain=voip.livewirenet.com
context=lwn
canreinvite=yes
Incoming calls from the peer lack an accountcode field.. a
NoOp(${ACCOUNTCODE}) is blank for calls coming from the above peer..
forcing an accountcode does indeed work with SetAccount() however, as I
stated, this does not scale.
I should also note that both incoming and outgoing calls involving the
peer (and many others) do not have an account code.. I am not concerned
with outgoing calls to the peer in this case.. just incoming from the peer..
I also saw in one list post that supposedly the accountcode entry in the
sip peer should not be relied upon.. but no detail or explanation was
given as to why that was..
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I should also expand this to note that I have also tested the amaflags variable and it also has no effect on my sip peer entries' cdr records. (latest stable asterisk has been tested) Matt Hess wrote:> My cdrs are missing accountcodes for incoming calls from other > asterisk servers.. > I've seen a few people mentioning this on the list and the solution > seems to be setting up a dialplan for incoming calls from a particular > sip peer.. in my opinion this does not scale well at all and I am > looking for a solution to correct this problem. > > example sip peer: > > [asterisk_gw] > type=friend > accountcode=asteriskgw > host=x.x.x.x > restrictcid=no > fromdomain=voip.livewirenet.com > context=lwn > canreinvite=yes > > Incoming calls from the peer lack an accountcode field.. a > NoOp(${ACCOUNTCODE}) is blank for calls coming from the above peer.. > forcing an accountcode does indeed work with SetAccount() however, as > I stated, this does not scale. > I should also note that both incoming and outgoing calls involving the > peer (and many others) do not have an account code.. I am not > concerned with outgoing calls to the peer in this case.. just incoming > from the peer.. > > I also saw in one list post that supposedly the accountcode entry in > the sip peer should not be relied upon.. but no detail or explanation > was given as to why that was.. > > >_______________________________________________ >Asterisk-Users mailing list >Asterisk-Users@lists.digium.com >http://lists.digium.com/mailman/listinfo/asterisk-users >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-------------- next part -------------- A non-text attachment was scrubbed... Name: mhess.vcf Type: text/x-vcard Size: 279 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20050721/dcb37084/mhess.vcf