Displaying 20 results from an estimated 400 matches similar to: "json.c:704 ast_json_vpack: Error building JSON from '{s: s, s: s}': Invalid UTF-8 string."
2015 Mar 10
2
json.c:704 ast_json_vpack: Error building JSON from '{s: s, s: s}': Invalid UTF-8 string.
05.03.2015 11:42, Dmitry Melekhov ?????:
> 05.03.2015 11:29, Dmitry Melekhov ?????:
>> Hello!
>>
>> Just installed asterisk 13.2.0 and see many such messages in log, I
>> see them in console during calls, really something like this:
>>
>>
>> -- Executing [6166 at kanbaikal:2] Dial("OOH323/kanbaikal-6",
>> "SIP/6166 at
2015 Mar 05
0
json.c:704 ast_json_vpack: Error building JSON from '{s: s, s: s}': Invalid UTF-8 string.
05.03.2015 11:29, Dmitry Melekhov ?????:
> Hello!
>
> Just installed asterisk 13.2.0 and see many such messages in log, I
> see them in console 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
2015 Mar 09
0
json.c:704 ast_json_vpack: Error building JSON from '{s: s, s: s}': Invalid UTF-8 string.
On Thu, Mar 5, 2015 at 2:29 AM, Dmitry Melekhov <dm at belkam.com> wrote:
> Hello!
>
> Just installed asterisk 13.2.0 and see many such messages in log, I see them
> in console during calls, really something like this:
>
>
> -- Executing [6166 at kanbaikal:2] Dial("OOH323/kanbaikal-6",
> "SIP/6166 at asterisk") in new stack
> == Using SIP
2015 Mar 10
0
json.c:704 ast_json_vpack: Error building JSON from '{s: s, s: s}': Invalid UTF-8 string.
On Tue, Mar 10, 2015 at 5:00 AM, Dmitry Melekhov <dm at belkam.com> wrote:
> 05.03.2015 11:42, Dmitry Melekhov ?????:
>
>> 05.03.2015 11:29, Dmitry Melekhov ?????:
>>>
>>> Hello!
>>>
>>> Just installed asterisk 13.2.0 and see many such messages in log, I see
>>> them in console during calls, really something like this:
>>>
2019 Dec 17
2
ARI strange bug on version 13.29.2
Hello,
I am using an ARI dialer for my applications and since my last upgrade
to Ver. 13.29.2 from 13.23.1 I am getting this strange bug from the ARI debugger:
Debugging on all applications enabled
<--- ARI request received from: x.x.x.x:63036 --->
HOST: x.x.x.x:8088
content-type: application/json
authorization: Basic xxxx
content-length: 265
body:
{
"context":
2016 Oct 17
4
Multiple readfile oddities, newlines etc
I have a plain text file, ASCII, unix line breaks. 1 single line, and all
that is in it is the word "radio".
Here's some test dialplan:
exten => 5,1,Verbose(Context: ${CONTEXT} Exten:${EXTEN})
same => n,Set(feature=${FILE(/home/test/feature-1.txt,0,1,l,u)})
same => n,Verbose(${feature})
same => n,Set(featurefile=/home/test/feature-1.txt)
same =>
2016 Oct 17
2
Multiple readfile oddities, newlines etc
On Tue, 18 Oct 2016, Pete Mundy wrote:
> If you want to know what is _really_ in that file (including all
> invisible characters and anything else that wc etc might not count),
> pipe it through 'hexdump'.
>
> cat?/home/test/feature-1.txt | hexdump
Or just:
hexdump /home/test/feature-1.txt
--
Thanks in advance,
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
2009 Oct 15
1
"Complex?" import of pdf files (criminal records) into R table
Hi there,
I'm facing the decision if it would be possible to transform several
more or less complex pdf files into an R Table-Format or if it has to be
done manually. I think it would be a impudent to expect a complete
solution, but I would be grateful if anyone could give me an advice on
how the structure of such a R-program could look like, and if it's
possible in general.
Here
2014 Jul 09
1
switching from simple_bridge technology to native_rtp issue
Hi,
with canreinvite=no and directmedia=no I and getting the message in the
logs for all calls
"switching from simple_bridge technology to native_rtp"
-- Executing [102 at mkg:1] Dial("SIP/101-00000017", "SIP/102") in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/102
-- SIP/102-00000018 is ringing
-- SIP/102-00000018 answered SIP/101-00000017
2014 Jan 10
1
asterisk 11.7.0: Delayed audio
On connection to an incoming call via PSTN where
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
2015 Mar 19
2
Asterisk switching bridge to native_rtp even with direct_media=no
NAT endpoint calling local endpount - switching to native_rtp then no audio, both of them have direct_media=no, Verbose log:
-- Executing [99 at dialmap:1] AGI("PJSIP/304-00000022", "/pbx/agi.php") in new stack
-- Launched AGI Script /pbx/agi.php
-- AGI Script Executing Application: (Dial) Options: (PJSIP/99/sip:99 at 192.168.1.73:5060,20)
-- Called
2015 Mar 25
2
TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 34
tnaks for your response but the number dialed exist and i can call this
number when i configure the trunk directly in x-lite and i call call also
this number from my cell phone .
any help
thanks and regards
2015-03-25 12:59 GMT+00:00 Matthew Jordan <mjordan at digium.com>:
> On Wed, Mar 25, 2015 at 7:35 AM, Salaheddine Elharit
> <salah.elharit200 at gmail.com> wrote:
> >
2017 Apr 18
4
Voicemail asking for login
On 2017-04-18 02:42 AM, Pete Mundy wrote:
> Try this:
>
> asterisk -r
> core set verbose 10
> [get user to trigger fault]
> [examine console output, and post to list if still unclear]
>
> If you don't solve it yourself, then we'll be able to help further once
> we've seen the output.
I can't see much more than at my previous debug level but here it is
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
2015 Mar 25
2
TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 34
hello list,
i have asterisk 11.15.0 and i have some trunks sip from my provider
we have some ip phone astra 6731i
each Ip-phone is configured with trunk and we call
no ihave configured another trunk from the same provider in my asterisk
i can call all numbers just the numbers are configured in thses ip phones.
but when i configured the same trunk in x-lite i can call theses ip-phones
without
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
2015 Aug 25
4
Ringback issue
My last problem was nicely solved through this mailing list so
hopefully this new problem will have the same happy outcome.
My situation is that I have many extensions. Here is a sample:
[client-phone](!)
type=friend
host=dynamic
secret=XXXXXXXXXX
dtmfmode=auto
disallow=all
allow=ulaw
allow=gsm
allow=g723
allow=ilbc
subscribemwi=no
[4165555555](client-phone)
secret=xxxxxxxxxxxxxxxxxxxxxxxxxx
2014 Jan 21
3
Asterisk Fax detection *11.7
Hello everybody
I'm trying to enable the Digium res_fax app at my *11.7 Server.
a fax show stats comes up with
FAX Statistics:
---------------
Current Sessions : 0
Reserved Sessions : 0
Transmit Attempts : 0
Receive Attempts : 1
Completed FAXes : 1
Failed FAXes : 1
Digium G.711
Licensed Channels : 1
Max Concurrent : 0
Success : 0
Switched to
2015 Mar 18
2
Asterisk switching bridge to native_rtp even with direct_media=no
Well, it breaks audio for all NAT endpoints, how can I fix this?
> On 18 Mar 2015, at 15:48, Matthew Jordan <mjordan at digium.com> wrote:
>
> On Wed, Mar 18, 2015 at 7:41 AM, Nick Awesome <jleed at me.com <mailto:jleed at me.com>> wrote:
>> Hey guys,
>>
>> have issues with reinvite, no matter what endpoint is calling asterisk
>> always tries