[Asterisk-Users] Using multiple iconnecthere accounts

[Martin Pycko]

How about that:

exten => _91NXXNXXXXXX,1,Dial,SIP/${EXTEN:1}@iconnect&SIP/${EXTEN:1}@iconnect2

Martin

On Tue, 8 Jul 2003, Derek Beaumont wrote:

> Asterisk has registered with both accounts:
>
> sip show registry
> Host                  Username     Refresh State
> 213.137.73.178:5060   xxxxxxxx         120 Registered
> 213.137.73.178:5060   xxxxxxxx         120 Registered
>
> I can make one call just fine, but when I try to make the second call,
> I get an invalid extension error.  When using the following
> configuration:
> >exten=>_91NXXNXXXXXX,1,StripMSD,1
> >exten=>_1NXXNXXXXXX,2,Dial,SIP/BYEXTENSION at iconnect
> >exten=>_1NXXNXXXXXX,3,Dial,SIP/BYEXTENSION at iconnect2
> I get the following output
>
> Executing Dial("Zap/4-1", "SIP/BYEXTENSION at iconnect") in new stack
>     -- Called xxxxxxxxxxx at iconnect
>     -- SIP/iconnect-cd45 is making progress passing it to Zap/4-1
>
> >show channels
> >Peer             Username    Call ID      Seq (Tx/Rx)  Lag      Jitter
> Format
> >213.137.73.178   xxxxxxxxxx  6631b1e766b  00103/00000  00000ms  0000ms
> 4
> >1 active SIP channel(s)
>
> This appears when I make the first call.  I notice that I have a 0ms
> Jitter buffer. I am now curious as to how I create a jitter buffer
> in sip.conf?  I have the following in the [general] section of sip.conf
>
> >jitterbuffer=yes
> >dropcount=3
> >maxjitterbuffer=2500
> >maxexccessbuffer=100
>
>
> Below is the output when I tried to call a second long distance number
>
>     -- Executing Dial("Zap/4-2", "SIP/BYEXTENSION at iconnect") in new
> stack
>     -- Called xxxxxxxxxxx at iconnect
>     -- Got SIP response 480 "Temporarily not available" back from
> 213.137.73.178
>     -- SIP/iconnect-fde9 is circuit-busy
>   == Everyone is busy at this time
>     -- Executing Dial("Zap/4-2", "SIP/BYEXTENSION at iconnect2") in new
> stack
>     -- Called xxxxxxxxxxx at iconnect2
> sip show channels
> Peer             Username    Call ID      Seq (Tx/Rx)  Lag      Jitter
> Format
> 213.137.37.178   xxxxxxxxxx  7047ee1a76b  00102/00000  00000ms  0000ms
> 2
> 213.137.73.176   xxxxxxxxxx  7b782a7b3dd  00103/00000  00000ms  0000ms
> 4
> 2 active SIP channel(s)
> *CLI> WARNING[98311]: File chan_sip.c, Line 410 (retrans_pkt): Maximum
> retries exceeded on call
> 7047ee1a76b10c2e56afddfe6bc01895 at xxx.xxx.xxx.xxx for seqno 102 (Request)
>   == No one is available to answer at this time
>     -- Sent into invalid extension 'xxxxxxxxxxx' in context 'outgoing'
> on Zap/4-2
>     -- Executing Playback("Zap/4-2", "TelError") in new stack
>     -- Playing 'TelError'
> WARNING[98311]: File chan_sip.c, Line 410 (retrans_pkt): Maximum retries
> exceeded on call 7047ee1a76b10c2e56afddfe6bc01895 at 67.70.231.220 for
> seqno 102 (Request)
>
> Any help is appreciated.  Thank you for your time.
>
> ========OLD MESSAGE===========================
>
> >>Did asterisk register with both accounts ?
> >>"sip show registry"
> >>
> >>Can you post what happens on the console along with 'sip debug' ?
> >>
> >>Martin
>
> On Tue, 8 Jul 2003, Derek Beaumont wrote:
>
> > Has anybody out there tried to use two different iconnecthere accounts
> > with Asterisk?
> > What I want to do is use a second account if the first is busy.
> > I have tried the following:
> >
> > exten=>_91NXXNXXXXXX,1,StripMSD,1
> > exten=>_1NXXNXXXXXX,2,Dial,SIP/BYEXTENSION at iconnect	;iconnect is the
> > first account
> > exten=>_1NXXNXXXXXX,3,Dial,SIP/BYEXTENSION at iconnect2	;iconnect2 is
> > the second account
> >
> > But that doesn't work.  Has anybody tried this before?
> >
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users at lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
>
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