[Asterisk-Users] Using multiple iconnecthere accounts

[Derek Beaumont]

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?
>
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