Ronald Wiplinger
2006-Apr-10 09:31 UTC
[Asterisk-Users] still no solution for me, if one provider fails.
I am still looking for a solution and I am sure that I am not the only
one having that problem:
If provider A fails for any reason, the next provider should be taken.
There are many reasons, why a provider fails, like:
password wrong (cli reports so, but actually it is the gateway's problem)
gateway temporary not reachable
gateway busy
...
Our user places a call, the gateway responds with no sound at all, or
hangs up, or gives busy tone.
How can we get to the next provider?
I have now:
exten => _9011Z.,103,Dial(SIP/011${EXTEN:4}@provider-a)
;exten => _9011Z.,103,Dial(SIP/011${EXTEN:4}@provider-b)
;exten => _9011Z.,103,Dial(SIP/011${EXTEN:4}@provider-c)
exten => _9011Z.,104,NoOp(${DIALSTATUS})
by hand I move the remark sign around!!!
How are you handling such situations?
bye
Ronald Wiplinger
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Derek Whitten
2006-Apr-10 09:45 UTC
[Asterisk-Users] still no solution for me, if one provider fails.
Ronald Wiplinger wrote:> I am still looking for a solution and I am sure that I am not the only > one having that problem: > > If provider A fails for any reason, the next provider should be taken. > > There are many reasons, why a provider fails, like: > password wrong (cli reports so, but actually it is the gateway's problem) > gateway temporary not reachable > gateway busy > ... > > Our user places a call, the gateway responds with no sound at all, or > hangs up, or gives busy tone. > > How can we get to the next provider? > > I have now: > exten => _9011Z.,103,Dial(SIP/011${EXTEN:4}@provider-a) > ;exten => _9011Z.,103,Dial(SIP/011${EXTEN:4}@provider-b) > ;exten => _9011Z.,103,Dial(SIP/011${EXTEN:4}@provider-c) > exten => _9011Z.,104,NoOp(${DIALSTATUS}) > > by hand I move the remark sign around!!! > > How are you handling such situations? > > bye > > Ronald Wiplinger > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-usershttp://www.voip-info.org/wiki/view/Asterisk+dial+plan+-+working+example about 1/3 down page.. look for [macro-dial-pstn-sip-iax] RTFW -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 189 bytes Desc: OpenPGP digital signature Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20060410/cb5277b4/signature.pgp