search for: 0123456789

Hello, Groups in asterisk are summarized here ( http://www.voip-info.org/wiki/view/Channels+and+Groups). Is there any difference between DAHDI/G1/0123456789 and DAHDI/g1/0123456789 (as I've been advised in another thread, to switch from one notation to the other and I can't see the reason behind that) ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/atta...

...I Tx >> agi_context: internos AGI Tx >> agi_extension: XXXXXXXXX AGI Tx >> agi_priority: 1 AGI Tx >> agi_enhanced: 0.0 AGI Tx >> agi_accountcode: XXXXXXXXXXX AGI Tx >> > AGI Rx << ANSWER AGI Tx >> 200 result=0 AGI Rx << STREAM FILE astcc-tone 0123456789 AGI Tx >> 200 result=0 endpos=11200 AGI Rx << STREAM FILE astcc-youhave 0123456789 AGI Tx >> 200 result=0 endpos=9920 AGI Rx << SAY NUMBER 6 0123456789 -- Playing 'digits/6' (language 'br') AGI Tx >> 200 result=0 AGI Rx << STREAM FILE astcc-dol...

I download and installed ASTCC over the weekend and I am having an issue where the INUSE flag will not get set back to 0 if the user drops a call while the balance is being played. All other times it seems to reset the flag correctly. I have tried both AGI and DeadAGI with the same results. Those of you using it for a while, how did you get around this? Just for fun this is all I am doing in

lvqcl wrote: > Oops. It seems that vsnprintf_s isn't always available on MinGW platform: > MinGW declares this function only if MINGW_HAS_SECURE_API macro is defined. > That's because WinXP version of msvcrt.dll doesn't contain secure functions > like vsnprintf_s. > > Maybe it's better to revert vsnprintf_s to vsprintf or to use vnsprintf? Ok, we need to drop

...same host 00010203-0405-0607-0809-0a0b0c0d0e0f if I type on both hosts: host1 virsh # sysinfo ....... <system> <entry name='manufacturer'>Supermicro</entry> <entry name='product'>X9SCL/X9SCM</entry> <entry name='version'>0123456789</entry> <entry name='serial'>0123456789</entry> <entry name='uuid'>00010203-0405-0607-0809-0A0B0C0D0E0F</entry> <entry name='sku'>To be filled by O.E.M.</entry> <entry name='family'>To be filled by O.E.M...

...use vnsprintf? > > Ok, we need to drop vsnprintf_s to support WinXP. I'd prefer vsnprintf > over vsprintf but have no way of testing any of these options. I wrote a small program that fills a buffer[] with "abcdefghijklmnopqrstuvwxyz\0" pattern and then tries to write "0123456789" string into it. It calls either ret = vsnprintf_s(buffer, buf_size, _TRUNCATE, fmt, va); or ret = vsnprintf(buffer, buf_size, fmt, va); The results are: --------------------------------------------------------------------------- vsnprintf_s (MSVC, MinGW): buf_size = 8; ret...

...message along the lines of "nosupported codecs in SDP" or somesuch. If anyone wants it, I can get the full message when I get home. When I try to use the DTA310, asterisk complains about authentication on SUBSCRIBE requests. Here is the relevant info in my sip.conf file: register =>0123456789@packet8.net/0123456789 ;the 0123456789 is my activation number for packet8 service... ;the DTA seems to want to use it. [packet8] type=friend disallow=gsm secret=mypassword allow=g723.1 ;expirey=15 sip_codec=g723.1 username=0403531400 fromuser=0403531400 host=packet8.net [0123456789] type=friend...

...tect)=yes/no to allow fax detection. It's done (see below) but still fax detection :-( Extension 300 is hylafax with iaxmodem. On the upper Asterisk gw it's the same, despite the faxdetect set to no we also have the NOTICE of T.38 re-INVITE. Test is done with a mobile phone calling the 0123456789 PSTN number. -- Executing [0123456789 at from-internal:1] Set("SIP/TOOTAi-00008262", "FAXOPT(faxdetect)=no") in new stack -- Executing [0123456789 at from-internal:2] Macro("SIP/TOOTAi-00008262", "Fax") in new stack -- Executing [s at macro-...

...ame call I can add some other custom headers (logs are below). Is there any chance I can rewrite Diversion header in this scenario with PJSIP_HEADER function? Asterisk version is 16.0.1 built from source on Debian 9. Thank you Davor # Outgoing context - TSP provider [outgoing] exten => 0123456789,1,Dial(PJSIP/${EXTEN}@${SBC_1},,b(add_diversion^FWDdiver^1)) same => n,Hangup() # Diversion manipulation context [add_diversion] exten => FWDdiver,1,Set(PJSIP_HEADER(add,Diversion)=<sip:full_pstn_no at example.com>\;reason=unconditional\;screen=yes\;privacy=off)) exten => FWDdiv...

...ction. > > It's done (see below) but still fax detection :-( Extension 300 is > hylafax with iaxmodem. > > On the upper Asterisk gw it's the same, despite the faxdetect set to no > we also have the NOTICE of T.38 re-INVITE. Test is done with a mobile > phone calling the 0123456789 PSTN number. > > -- Executing [0123456789 at from-internal:1] > Set("SIP/TOOTAi-00008262", "FAXOPT(faxdetect)=no") in new stack > -- Executing [0123456789 at from-internal:2] > Macro("SIP/TOOTAi-00008262", "Fax") in new stack >...

...; Answer the line exten => s,n,DigitTimeout,5 ; Set Digit Timeout to 5 seconds exten => s,n,ResponseTimeout,10 ; Set Response Timeout to 10 seconds .and so on. That's all I have.have I missed something? Debug output from call: 192.1.1.1=my server 0123456789=my number at SIP-provider 9999999999=the number I'm calling from 213.132.103.213, 212.112.162.50=my SIP providers IPs ========================================== Sip read: INVITE sip:s@192.1.1.1 SIP/2.0 Record-Route: <sip:213.132.103.213:5060;transport=UDP;lr=true> Via: SIP/2.0/UDP...

...22222233333333333333344444+61 9 | 00000000000001111111111111222222222223333333333444444444555555556666+14 10 | 00000001111111222222233333334444444555555666666777777888888999999 11 | 00000111112222233333444445555566667777888999 12 | 000111222333444555666777888999 13 | 0011223456789 14 | 0123456789 15 | 0123456789 16 | 0123456789 17 | 0123456789 18 | 01 > (sorry for the poor line wrapping) Finally, the quantile() command works fine directly pasted onto the command line: > quantile(last.hc.actors,probs=seq(0,1,0.1),na.rm=T) 0% 10% 20% 30% 40% 50% 60% 70% 80% 90% 1...

...AHDI span, must have CallerIDs presented without any prefix. Ideally, CallerID should be presented : 1- with 4-digits for internal phones 2- with 10-digits for external phones so that both phones can return the call without re-dialing. Suggestions ? A is 1234 alias DID 0555551234 B is 5678 C is 0123456789 I was thinking of using something like this: Dial(SIP/5678<option_to_present_1234_to_callee>&DAHDI/g1<option_to_present_0555551234>/0123456789) What could be <option_to_present_1234_to_callee> and <option_to_present_0555551234> Regards -------------- next part -------...

...channel # # # #################################### sub send_file { my ($myfile) = @_; chomp($myfile); if ($DEBUG == 1 ) { print DEBUGOUT "Sending stream $myfile \n"; } print "STREAM FILE $myfile \"0123456789\"\n"; my $result = <STDIN>; &checkresult($result); } ############ hangup ############### # Use this to hand up a channel # # the channel # # # #################################### sub hangup { if ($DEBUG == 1...

...xtension [exten@][context] This will originate a call between the specified channel tech/data and the given extension. If no context is specified, the 'default' context will be used. If no extension is given, the 's' extension will be used." I would like for example to call 0123456789 number from SIP/7530 extension. My asterisk server is set to use "local" context for outgoing calls. My first idea was to type this : originate SIP 7530 0123456789 at local But it fails : it keeps displaying " There are two ways ..." and nothing else seem to occur. Can anyo...

Hello, Did someone have a solution for a line fax detection for outgoing call For exemple I call number 0123456789 - if it is a fax then redirect to extension A - if it is a line then redirect to exention B whats ia want its somthing like AMD application that i use for the answering machine . http://www.voip-info.org/wiki/view/Asterisk+cmd+AMD search in the wiki give this application : http://www.voip-info...

Hi Steve, What language is that please? We're using Perl and so far I haven't found an equivalent there. Thanks for your help. On Fri, 7 Jun 2019 at 12:10, Steve Edwards <asterisk.org at sedwards.com> wrote: > On Fri, 7 Jun 2019, David Cunningham wrote: > > > We have a need to record audio and allow the user to press any DTMF key > > to end the recording.

...simultaneousMax=10 language=de ; erweitertes logging aktivieren (debugging) wrapLibTraceLevel=9 libTraceLevel=9 libTraceFile=/var/log/asterisk/oh323.log ; gatekeeper des carrier gatekeeper=XXX.XXX.XXX.XXX gatekeeperTTL=600 userInputMode=TONE ; detailierte cdr erstellen amaFlags=billing accountCode=0123456789 ; eingehende calls an diesen context senden context=carrier-in [register] context=carrier-in alias=0123456789 [codecs] codec=G711A frames=20 2.) Status of OpenH323 channel driver --------------------------------------- *CLI> oh323 show conf Version: 0.6.6 Listening on address: 0.0.0.0:17...

...; Answer the line exten => s,n,DigitTimeout,5 ; Set Digit Timeout to 5 seconds exten => s,n,ResponseTimeout,10 ; Set Response Timeout to 10 seconds and so on That?s all I have have I missed something? Debug output from call: 192.1.1.1=my server 0123456789=my number at SIP-provider 9999999999=the number I?m calling from 213.132.103.213, 212.112.162.50=my SIP providers IPs ========================================== Sip read: INVITE sip:s@192.1.1.1 SIP/2.0 Record-Route: <sip:213.132.103.213:5060;transport=UDP;lr=true> Via: SIP/2.0/UDP 213....

Hi, I m trying to use perl script to generate call with a server asterik . If I use telnet session to generate like this : $telnet localhost 5038 Action: Login Username: useroperator Secret: password Action: Originate Context: context Channel: Local/0123456789 at context Exten: 221 Priority: 1 it works good :) instead with a script perl like this : .... use Net::Telnet (); .... $tn->print("Action: Login\nUsername:$USERNAME\n Secret:$SECRET\n\n"); $tn->waitfor('/Authentication accept*/') $tn->print("Action: Origin...