search for: dumpchan

Hi Daniel, Am 27.03.20 um 09:24 schrieb Administrator: > Hangup is h extension. your macro will never be executed. Solution: > > same = n,Dial(whatever) > same = n,[...]) > same = n,Hangup > > exten  = h,1,1,DumpChan() >  same = n,System(/home/asterisk/bash_test) I don't really understand your code… I think I don't have to edit the first part of the conf file (" same = n,Dial(whatever) "), you just mean the second part of the code is executed by "n,Hangup"? Then I have to add th...

...every-inbound-hangup-call/45169 [2] https://community.freepbx.org/t/email-notification-of-incoming-missed-call/29913 -------------- next part -------------- #!/bin/bash cd ~asterisk echo test > /home/asterisk/test -------------- next part -------------- [macro-hangupcall-custom] exten => s,1,DumpChan() exten => s,n,System(/home/asterisk/bash_test) exten => s,n,MacroExit()

...m that my SIP service provide does actually pass on those information? Here is what I have in extensions.conf to test this scenario exten => _XX.,1,NoOp(Call received from VoicePulse) exten => _XX.,n,Log(INFO|Caller ID Number: ${CALLERID(num)}) exten => _XX.,n,Answer() exten => _XX.,n,DumpChan() exten => _XX.,n,VoiceMail(101 at default,u) Here is what I see on the console. zeus*CLI> -- Executing [12222222 at voicepulse-in:1] NoOp("SIP/mrXXXX-08XXXX", "Call received from VoicePulse") in new stack -- Executing [12222222 at voicepulse-in:2] Log("SIP/...

It seems that what ever I set in my answer handler does not show up in the hangup handler. In order to do billing I can't rely on the g option where the caller hangs up the call. Looks like I can either use h or a hangup handler along with the shared function. On Tue, Dec 26, 2017 at 4:40 PM, Eric Wieling <ewieling at nyigc.com> wrote: > Don't use an 'h' extension, use

...19, Kai Herlemann a écrit : > Hi Daniel, > > Am 27.03.20 um 09:24 schrieb Administrator: >> Hangup is h extension. your macro will never be executed. Solution: >> >> same = n,Dial(whatever) >> same = n,[...]) >> same = n,Hangup >> >> exten  = h,1,1,DumpChan() >>  same = n,System(/home/asterisk/bash_test) > I don't really understand your code… > > I think I don't have to edit the first part of the conf file (" same = > n,Dial(whatever) "), you just mean the second part of the code is > executed by "n,Hangup&...

-- SIP/ vaso -e26c answered Zap/14-1 -- Executing DumpChan("SIP/ vaso -e26c", "") in new stack -- Executing DumpChan("SIP/vaso-e26c", "") in new stack Dumping Info For Channel: SIP/vaso-e26c: ============================================================================ ==== Info: Name=...

...ransfer is > > performed. > > > > Is this a bug or is there something that needs to be set to allow the > > DYNAMIC_FEATURES to be inherited after an attended transfer from a queue? > > Doing some more tests, this reads like a bug to me. > Using a hanguphandler with DumpChan in the dialplan context that executes > the Queue, I can see that DYNAMIC_FEATURES is set. > After the attended transfer when the call is ended, the hanguphandler > still shows that DYNAMIC_FEATURES is set. It's just not accessible. > > Any thoughts? > It likely depens on how...

...n't have the /usr/lib/asterisk/modules/codec_g729a.so file in place it works. I use or used to use the licensed G729 Codec from Digium. This is the error message from asterisk -vvg: [app_playback.so] => (Sound File Playback Application) == Registered application 'Playback' [app_dumpchan.so] => (Dump Info About The Calling Channel) == Registered application 'DumpChan' [app_zapateller.so] => (Block Telemarketers with Special Information Tone) == Registered application 'Zapateller' [codec_ilbc.so] => (iLBC/PCM16 (signed linear) Codec Translator) == Re...

...NEL})}) same => n,Dial(SIP/${CLIENT}@1.1.1.1:5080,,gU(verify_human)) [verify_human] Exten => s,1,Noop() same => n,MixMonitor(/tmp/TO_CALL_CENTER_${uid}-${EPOCH}_SECOND.WAV) same => n,AMD same => n,ExecIf($["${AMDSTATUS}" != "FOUND"]?Hangup(16))) same => n,DumpChan() -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20180713/decdd27a/attachment.html>

Hi all First I want to wish for everone a happy new year... Well... I have run asterisk 1.4.16.1 in a server. I have this IVR, in extensions.conf: [ura] ;exten => s, 1, Wait,1 exten => s, 1, Answer() exten => s, n, Noop() exten => s, n(debug),DumpChan() exten => s, n, Set(LANGUAGE()=pt_BR) exten => s, n, Set(CALLFILENAME=/var/spool/asterisk/monitor/entrada/) exten => s, n, Set(DYNAMIC_FEATURES = hangup#pickupexten#atxfer#blidxfer) exten => s, n, MixMonitor(${CALLFILENAME}/${STRFTIME(${EPOCH},,%d%m%Y-%H:%M:%S)}-${EXTEN}.wav|v(0)V(0))...

hi list can U help me caller id in USTM if now working -- Starting switch on '4211 at 4211-1' to 4203 -- Executing [4203 at office:1] DumpChan("USTM/4211 at 4211-0x7f7ba4228fd0", "") in new stack Dumping Info For Channel: USTM/4211 at 4211-0x7f7ba4228fd0: ================================================================================ Info: Name= USTM/4211 at 4211-0x7f7ba4228fd0 Type= USTM...

I am trying to use ImportVar to get some information out of a SIP/ZAP channel. I cannot seem to find an example of the syntax, or what variables I can access. Basically, I would like to output which person is being called. i.e: SIP/25 calls SIP/21. 25 executes a macro, and the result is SIP/21. The info that I want is stored in the channel's "Direct Bridge" variable. I have

Hi, I've just discovered IMPORT function existence. It can be use to import values from channel's Variable section but unfortunately, il can't be use to access to values from Info section (I'm referring here to sections Info and Variables dumped by DumpChan application). Is there a way to work around this and access from one channel for instance to another channel's CallerIDNum variable ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090615/9...

...d to channel C and B1 and B2 are hung up. In the h extension for channel B2, the channel is renamed to B2<ZOMBIE> and i see that the channel variables of A have been merged into B2<ZOMBIE>. If there were duplicate names for variables, the channel now has those variables doubled. The DumpChan() application called from the h extension confirms this. In my case the channels are all SIP channels and in the h extension I want to access the SIPCALLID variable of the A channel. Every access to it gives me the wrong value namely that of channel B1. How do i access the _second_ variable nam...

...ransfer = yes echocancel = yes echotraining = yes immediate = no context = longdistance signalling = fxo_ks [test1](phone_template) callerid = "Test 1" <(111)222-3333> setvar=myvariable=test dahdichan = 1 I have tried every example I have been able to find but nothing appears in a DumpChan. Thank you. Chet Stevens -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20121201/13f68435/attachment.htm>

Is there a way to list the names of the channel variables that are currently defined on a given channel via AMI? I know of GetVar and SetVar, but GetVar needs the name of the variable to get.

...riority: 1 Async: true Variable: CHANNEL_TO_CUSTOMER=SIP/voipms/1112223333 Timeout: 999999 Dial Plan: [callmenow] exten => s,1,NoOp(callmenow: Queue without answer) same =>n,Queue(sales,Rtc) [dial-to-customer] exten => s,1,NoOp(dial-to-customer channel=${CHANNEL(name)}) same =>n,DumpChan() The dial-to-customer context is invoked when the sales queue agent answers the phone. When the local channel is used, the queue related variables, specifically MEMBERINTERFACE, are missing. When a normal call (typically SIP or DAHDI channel) enters the queue, the MEMBERINTERFACE and other...

Hi, I think I've identified an issue and just want to check before completing a bug report. Prior to a call entering a Queue, I set __DYNAMIC_FEATURES=NewRecordApp. AgentA answers and is able to use that feature code. If AgentA performs an attended transfer of a call from a queue to AgentB, the feature code no longer works. Cases that do work are as follows... Calls using both Queue() and

...ving end (dial-dest in the example above) has become deaf! I've turned on debug and verbose to level 9, and there's nothing. It connects, starts music on hold, and then just ignores everything. Anything else I can add to the dialplan to see what might be causing this? (I've also tried dumpchan, too). It USED to work, and some point in the last week, it stopped working. (But the test dialplan above works). Mind boggled! Just to double check, yes, it's all set OK features show Builtin Feature Default Current --------------- ------- ------- Pickup...

On 15-09-28 10:19 AM, Emil Ohlsson wrote: > (Still no not receiving the mail, revisited the settings.) > > OK, so SendText doesn't work with this scenario. But can MessageSend > handle this, and respond even when the transport protocol is TLS? Or > do I need to modify Asterisk to add this support? MessageSend has no concept of TLS, it gets passed to chan_sip which then sends