similar to: 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

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

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

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: >>>

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":

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 =>

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,

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

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

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

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

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

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: > >

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

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

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

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

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

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

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