Displaying 5 results from an estimated 5 matches for "0000010d".
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2015 Mar 20
2
outbound calls
i noticed that when i active the voicemail in the IP-phone where the number
0033149xxxxxx is configured i can call this number without issue
Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called SIP/FD/0033149xxxxxx == Begin MixMonitor Recording
SIP/101-0000010d
-- SIP/FD-0000010e is making progress passing it to SIP/101-0000010d
> 0x2b393cfc2610 -- Probation passed - setting RTP source address to
192.
168.1.138:55542
> 0x1d08efa0 -- Probation passed - setting RTP source address to
217.195.xx.xx:46346
-- SIP/FD...
2015 Mar 20
0
outbound calls
...noticed that when i active the voicemail in the IP-phone where the
> number 0033149xxxxxx is configured i can call this number without issue
>
> Using SIP RTP TOS bits 184
> == Using SIP RTP CoS mark 5
> -- Called SIP/FD/0033149xxxxxx == Begin MixMonitor Recording
> SIP/101-0000010d
> -- SIP/FD-0000010e is making progress passing it to SIP/101-0000010d
> > 0x2b393cfc2610 -- Probation passed - setting RTP source address
> to 192.
> 168.1.138:55542
> > 0x1d08efa0 -- Probation passed - setting RTP source address to
> 2...
2015 Mar 20
0
outbound calls
...3149xxxxxx is configured i can call this number without issue
the server asterisk and the ip-phone where the number is configured are in
the same network 192.168.1.X
Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called SIP/FD/0033149xxxxxx
== Begin MixMonitor Recording SIP/101-0000010d
-- SIP/FD-0000010e is making progress passing it to SIP/101-0000010d
> 0x2b393cfc2610 -- Probation passed - setting RTP source address to
192.
168.1.138:55542
> 0x1d08efa0 -- Probation passed - setting RTP source address to
217.195.xx.xx:46346
-- SIP/FD...
2010 Jul 02
1
Transfer fails
...ected.
When I call the extension 20 directly from SIPaccount test1, the CLI
shows no problem :
[Jul 2 10:55:02] -- Executing [20 at from-TEST:1]
Dial("SIP/test1-0000010c", "SIP/test2") in new stack
[Jul 2 10:55:02] -- Called test2
[Jul 2 10:55:02] -- SIP/test2-0000010d is ringing
So why can I call extension 20 (test2) directly but not transfer a call
to it ??
Jonas.
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2015 Mar 20
3
outbound calls
hello list
i have an issue related to outbound calls i can contact all the number
except on number given by our provider in trunk
the issue just when i configure my trunk in our server but when i configure
the trunk directly in x-lite i can contact this number without issue
below the cli
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Executing [0149xxxxxx at