Displaying 20 results from an estimated 63 matches for "probation".
2015 Mar 05
4
json.c:704 ast_json_vpack: Error building JSON from '{s: s, s: s}': Invalid UTF-8 string.
...during calls, really something like this:
-- Executing [6166 at kanbaikal:2] Dial("OOH323/kanbaikal-6",
"SIP/6166 at asterisk") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called SIP/6166 at asterisk
> 0x7fa9d4007660 -- Probation passed - setting RTP source
address to 10.18.0.19:26052
-- SIP/asterisk-0000000c is making progress passing it to
OOH323/kanbaikal-6
-- SIP/asterisk-0000000c is ringing
> 0x7fa9d4007660 -- Probation passed - setting RTP source
address to 10.18.0.19:26052
> 0x7fa9d...
2015 Mar 20
2
outbound calls
...xxxxx 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-0000010e answered SIP/101-0000010d
> 0x1d08efa0 -- Probation passed - setting RTP source address to
2...
2015 Mar 10
2
json.c:704 ast_json_vpack: Error building JSON from '{s: s, s: s}': Invalid UTF-8 string.
...gt; -- Executing [6166 at kanbaikal:2] Dial("OOH323/kanbaikal-6",
>> "SIP/6166 at asterisk") in new stack
>> == Using SIP RTP TOS bits 184
>> == Using SIP RTP CoS mark 5
>> -- Called SIP/6166 at asterisk
>> > 0x7fa9d4007660 -- Probation passed - setting RTP source
>> address to 10.18.0.19:26052
>> -- SIP/asterisk-0000000c is making progress passing it to
>> OOH323/kanbaikal-6
>> -- SIP/asterisk-0000000c is ringing
>> > 0x7fa9d4007660 -- Probation passed - setting RTP source
>&g...
2009 Oct 15
1
"Complex?" import of pdf files (criminal records) into R table
...ce since: xx.xx.1902
Date of offense:xx.xx.xxxx
Elements of the offence: For example "Rape"
Section in law: ?176, ?178 Abs. 1
Sentenced to 5 years imprisonment
"Irrelevant text for us"
Accommodation in an forensic psychiatry
Accommodation sentenced on probation
Rest of sentence sentenced on probation until the xx.xx.xxxx
2. xx.xx.1910
Be in force since: ....
.....
-----------------------------------------------------------------------
The problem is that the entries do not always have the same structure.
The first 6 lines are structurally...
2014 Jul 09
1
switching from simple_bridge technology to native_rtp issue
...2e3a9-e4be-4b2b-bf55-0357dafcdbab>
-- Channel SIP/102-00000018 joined 'simple_bridge' basic-bridge
<0ad2e3a9-e4be-4b2b-bf55-0357dafcdbab>
> Bridge 0ad2e3a9-e4be-4b2b-bf55-0357dafcdbab: switching from
simple_bridge technology to native_rtp
> 0x7f427c068a10 -- Probation passed - setting RTP source address to
111.118.250.236:49344
> 0x7f427c068a10 -- Probation passed - setting RTP source address to
111.118.250.236:49344
> 0x7f42500168d0 -- Probation passed - setting RTP source address to
111.118.250.236:26326
> 0x7f42500168d0 -- Probat...
2014 Jan 10
1
asterisk 11.7.0: Delayed audio
...ere
asterisk [11.7.0] is Dialing an internal extension
on answering the call there is about 6-7 seconds before
audio is heard on either side.
When looking at the CLI traces when I answer the incoming call that
asterisk extensions were dialing, I see immediately upon answering
>0xhexnumber -- Probation passed - setting RTP source address to
192.168.1.11:portnumber
then not until about 6 seconds later I see this
>0xhexnumber -- Probation passed - setting RTP source address to
192.168.1.11:diffportnumber
and immediately hear audio
what appears to be an issue is that the RTP link(audio) setup...
2015 Mar 19
2
Asterisk switching bridge to native_rtp even with direct_media=no
...65bf8e5e28: switching from simple_bridge technology to native_rtp
> Locally RTP bridged 'PJSIP/99-00000023' and 'PJSIP/304-00000022' in stack
> Locally RTP bridged 'PJSIP/99-00000023' and 'PJSIP/304-00000022' in stack
> 0x7f4b50145420 -- Probation passed - setting RTP source address to 194.204.157.200:8972
> 0x7f4b5014f140 -- Probation passed - setting RTP source address to 192.168.1.73:5004
-- Channel PJSIP/304-00000022 left 'native_rtp' basic-bridge <da8840bc-9b71-4ca6-b1d8-9565bf8e5e28>
-- Channel PJSIP/99-...
2015 Mar 05
0
json.c:704 ast_json_vpack: Error building JSON from '{s: s, s: s}': Invalid UTF-8 string.
...ke this:
>
>
> -- Executing [6166 at kanbaikal:2] Dial("OOH323/kanbaikal-6",
> "SIP/6166 at asterisk") in new stack
> == Using SIP RTP TOS bits 184
> == Using SIP RTP CoS mark 5
> -- Called SIP/6166 at asterisk
> > 0x7fa9d4007660 -- Probation passed - setting RTP source
> address to 10.18.0.19:26052
> -- SIP/asterisk-0000000c is making progress passing it to
> OOH323/kanbaikal-6
> -- SIP/asterisk-0000000c is ringing
> > 0x7fa9d4007660 -- Probation passed - setting RTP source
> address to 10.18.0.19...
2015 Mar 09
0
json.c:704 ast_json_vpack: Error building JSON from '{s: s, s: s}': Invalid UTF-8 string.
...ike this:
>
>
> -- Executing [6166 at kanbaikal:2] Dial("OOH323/kanbaikal-6",
> "SIP/6166 at asterisk") in new stack
> == Using SIP RTP TOS bits 184
> == Using SIP RTP CoS mark 5
> -- Called SIP/6166 at asterisk
> > 0x7fa9d4007660 -- Probation passed - setting RTP source address to
> 10.18.0.19:26052
> -- SIP/asterisk-0000000c is making progress passing it to
> OOH323/kanbaikal-6
> -- SIP/asterisk-0000000c is ringing
> > 0x7fa9d4007660 -- Probation passed - setting RTP source address to
> 10.18.0.19:26...
2015 Mar 25
2
TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 34
...n the same network
> > 192.168.1.x
> >
> > == Using SIP RTP TOS bits 184
> > == Using SIP RTP CoS mark 5
> > -- Called SIP/FD/0033149XXXXXX
> > -- SIP/FD-000000b9 is making progress passing it to SIP/306-000000b8
> > > 0x2afec424c430 -- Probation passed - setting RTP source address
> to
> > 192.168.1.212:57592
> > > 0xc5922b0 -- Probation passed - setting RTP source address to
> > 217.195.xx.xxx:29674
> > -- Got SIP response 556 "No address found" back from
> 217.195.XX.XXX:5060
> >...
2015 Mar 10
0
json.c:704 ast_json_vpack: Error building JSON from '{s: s, s: s}': Invalid UTF-8 string.
...g [6166 at kanbaikal:2] Dial("OOH323/kanbaikal-6",
>>> "SIP/6166 at asterisk") in new stack
>>> == Using SIP RTP TOS bits 184
>>> == Using SIP RTP CoS mark 5
>>> -- Called SIP/6166 at asterisk
>>> > 0x7fa9d4007660 -- Probation passed - setting RTP source address
>>> to 10.18.0.19:26052
>>> -- SIP/asterisk-0000000c is making progress passing it to
>>> OOH323/kanbaikal-6
>>> -- SIP/asterisk-0000000c is ringing
>>> > 0x7fa9d4007660 -- Probation passed - setting...
2017 Apr 18
4
Voicemail asking for login
...2 at LocalSets:19] Set("SIP/alex-00000175",
"_ACCOUNT=stocktrans2") in new stack<<<
-- Executing [stocktrans2 at LocalSets:20]
VoiceMail("SIP/alex-00000175", "stocktrans2 at VoiceMail,u") in new stack<<<
> 0x7f7fea5dc000 -- Probation passed - setting RTP source
address to 72.143.94.110:28503<<<
-- <SIP/alex-00000175> Playing
'/var/spool/asterisk/voicemail/VoiceMail/stocktrans2/unavail.gsm'
(language 'en')<<<
> 0x7f7fea5dc000 -- Probation passed - setting RTP source
add...
2015 Mar 20
0
outbound calls
...s 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-0000010e answered SIP/101-0000010d
> > 0x1d08efa0 -- Probation passed -...
2014 May 07
0
Video with asterisk12 and pjsip
Hi,
I tried to turn on Video and get the following cli-WARNING output
-- Executing [8600 at outgoing-kamailio:1] Answer("PJSIP/7000-00000000",
"") in new stack
> 0x7f46f41ff2e0 -- Probation passed - setting RTP source address to
192.168.8.203:17200
-- Executing [8600 at outgoing-kamailio:2]
ConfBridge("PJSIP/7000-00000000", "8600") in new stack
-- <PJSIP/7000-00000000> Playing 'conf-onlyperson.g722' (language 'de')
-- <PJSIP/7000-00000000...
2015 Mar 20
0
outbound calls
...mber 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-0000010e answered SIP/101-0000010d
> 0x1d08efa0 -- Probation passed - setting RTP source address to
2...
2015 Mar 25
2
TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 34
...nd i use
extension
all the ip-phone and x-lite and server asterisk in the same network
192.168.1.x
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called SIP/FD/0033149XXXXXX
-- SIP/FD-000000b9 is making progress passing it to SIP/306-000000b8
> 0x2afec424c430 -- Probation passed - setting RTP source address to
192.168.1.212:57592
> 0xc5922b0 -- Probation passed - setting RTP source address to
217.195.xx.xxx:29674
-- Got SIP response 556 "No address found" back from 217.195.XX.XXX:5060
== Everyone is busy/congested at this time (1:0/1/0)...
2015 Mar 20
3
outbound calls
...ialout-trunk:22] Dial("SIP/101-00000103",
"SIP/FD/0033149xxxxxx,300,") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called SIP/FD/0033149xxxxxx
-- SIP/FD-00000104 is making progress passing it to SIP/101-00000103
> 0x1d47d800 -- Probation passed - setting RTP source address to
192.168.1.138:54690
> 0x1d4faf90 -- Probation passed - setting RTP source address to
217.195.xx.xxx:36928
-- Got SIP response 556 "No address found" back from 217.195.xx.xxx:5060
== Everyone is busy/congested at this time (1:0/1/0)...
2015 Mar 19
0
Asterisk switching bridge to native_rtp even with direct_media=no
...; simple_bridge technology to native_rtp
> > Locally RTP bridged 'PJSIP/99-00000023' and 'PJSIP/304-00000022' in
> stack
> > Locally RTP bridged 'PJSIP/99-00000023' and 'PJSIP/304-00000022' in
> stack
> > 0x7f4b50145420 -- Probation passed - setting RTP source address to
> 194.204.157.200:8972
> > 0x7f4b5014f140 -- Probation passed - setting RTP source address to
> 192.168.1.73:5004
> -- Channel PJSIP/304-00000022 left 'native_rtp' basic-bridge
> <da8840bc-9b71-4ca6-b1d8-9565bf8e5e28>...
2015 Mar 29
0
Help! How to make Asterisk support ICE in public network
...native_rtp
> Remotely bridged 'SIP/6003-00000001' and 'SIP/6004-00000000' - media will flow directly between them
> Remotely bridged 'SIP/6003-00000001' and 'SIP/6004-00000000' - media will flow directly between them
> 0x7f5968006760 -- Probation passed - setting RTP source address to 114.81.254.172:4145
> 0x1fefbb0 -- Probation passed - setting RTP source address to 114.92.58.65:7076
st-srv-cs2*CLI>
st-srv-cs2*CLI>
st-srv-cs2*CLI>
-- Channel SIP/6004-00000000 left 'native_rtp' basic-bridge <2a01fb30-96e...
2014 Mar 24
1
Asterisk 11.8.0 and 11.8.1
I have used every asterisk 11.8.X version.
Have not had an issue till 11.8.0 and 11.8.1
When I use ConfBridge I am attempting to put all
participants in MUTE mode and just one talker...
However, since 11.8.0 I am hearing feedback in the
announcement like the channel is not really muted.
I dropped back to 11.7.0 and I hear no feedback.
Has something changed - or - am I not correctly setting
up