search for: ramais

Hi! I tried to use Parking Calls I use Asterisk 13, but I can't park any calls and it returns me [Feb 3 16:56:11] WARNING[1693]: pbx.c:12543 ast_context_verify_includes: Context 'ramais' tries to include nonexistent context 'parkedcalls' What is the correct code for put in extensions.conf? Can be this example below? [parkedcalls] exten => 700,1,ParkedCall(701) exten => 702,1,ParkedCall(702) exten => 703,1,ParkedCall(703) exten => 704,1,ParkedCall(704) I...

...azuco at gmail.com> > wrote: > >> Hi! >> >> I tried to use Parking Calls >> >> I use Asterisk 13, but I can't park any calls and it returns me >> >> [Feb 3 16:56:11] WARNING[1693]: pbx.c:12543 >> ast_context_verify_includes: Context 'ramais' tries to include >> nonexistent context 'parkedcalls' >> > > Are you loading res_parking.so? > > Does your res_parking.conf define a parkext and specify the context? > Documented in configs/samples/res_parking.conf.sample: > parkext => 700...

Ah no, I'm asking what code I put inside of parkedcalls? This example works? [ramais] include => parkedcalls [parkedcalls] exten => 700,1,ParkedCall(701) exten => 702,1,ParkedCall(702) exten => 703,1,ParkedCall(703) exten => 704,1,ParkedCall(704) This exten works? 2016-02-03 17:27 GMT-02:00, Doug Lytle <support at drdos.info>: >>>> On Feb 3, 20...

...NREACHABLE 1 iax2 peers [0 online, 1 offline, 0 unmonitored] See my both iax.conf SERVER 1 [General] bindport=4659 bindaddr = all disallow=all allow=ulaw;alaw ;Contas para os servidores das filiais. ;; [prote1-prote2] secret= password username=prote1-prote2 host=192.168.25.26 type=friend context=ramais qualify=yes trunk=yes auth = md5 ;; SERVER 2 [General] bindport=4659 bindaddr=all disallow=all allow=ulaw;alaw ;Contas para os servidores das filiais. ;; [prote1-prote2] secret= password username=prote1-prote2 host=192.168.25.25 type=friend context=ramais qualify=yes trunk=yes auth = md5 ;; ;;...

...-> Multiple warning messages 139.... 150 Dropping incompatible voice frame on Local/ D attended Transfer to E -> Multiple warning messages 139.... 150 Dropping incompatible voice frame on Local/ ... [Nov 10 16:12:38] NOTICE[4463] channel.c: Dropping incompatible voice frame on Local/413 at ramais-administrativo-81ea,2 of format slin since our native format has changed to 0x8 (alaw) ... [Nov 10 16:12:38] NOTICE[4463] channel.c: Dropping incompatible voice frame on Local/413 at ramais-administrativo-81ea,2 of format slin since our native format has changed to 0x8 (alaw) Can it cause a proble...

Hello! I'm using Asterisk for our office telephony, but we have some problems that still we can't resolve about it. Here they are: 1) merge in/out call recording files I also tried to use a script I found on the internet, called 2wav2mp3 In extensions.conf I added the following lines ; script to be executed when monitoring has been finished MONITOR_EXEC=/usr/local/bin/2wav2mp3 exten

..., Queue(8600|tT|||30) exten => 2, 1, Set(LANGUAGE()=pt_BR) exten => 2, 2, Queue(8500|tT|||30) exten => i, 1, Wait,1 exten => i, 2, Playback(invalid) exten => i, 3, Goto(s,11) exten => t, 1, Wait,1 exten => t, 2, Dial(SIP/8024,150,tTrwW) exten => s, n, HangUp() ;include => ramais Well, when I atempt dial to a external number, I goto this IVR. When the user get a connection, and when he press any invalid number, I get this message on CLI: Invalid extension '3' in context 'ura' on SIP/bitmixinfo-0825f158 Here, we suppost tha the user press key 3, for a mist...

...f I do the call through ORIGINATE Action, the Monitor() just makes the record file, but this is empty, if I do the call to SoftPhone the record file is made normally. My Action Originate: Action: Originate ActionID: 1BV2bwdI_#Ps20070620175008 Channel: Local/09194**** Exten: 101 Context: ramais Priority: 1 Variable: ACTIONID=1BV2bwdI_#Ps20070620175008 Async: True Part of my dial plan: exten => _09194****,1,set(SCREEN_FILE=/calls/records/TESTE_${RAND(1,99999)} exten => _09194****,2,Monitor(wav,${SCREEN_FILE},m) exten => _09194****,3,Dial(SIP/TmaisMG/9194****,40,rtw)...

...sip.conf file, everything works great. My extconfig.conf: sipusers => pgsql,asterisk,sip sippeers => pgsql,asterisk,sip voicemail => pgsql,asterisk,voicemail My configuration from 4111 stored in the Postgres: callgroup = 2 callerid = Ramal 4111 <4111> canreinvite = yes context = ramais host = dynamic nat = yes pickupgroup = 2 qualify = yes secret = 123456 type = friend username = 4111 disallow = all allow = alaw cancallforward = yes call-limit = 6 My general section of sip.conf : [general] qualify=yes context=default allowoverlap=no bindport=5060 bindaddr=0.0.0.0 srvlookup=yes...

Hey there, I am trying to setup a2billing on asterisk 1.6 , but ,when I try to access its web page I see the a2billing directories:Index of /a2billingNameLast modifiedSizeDescriptionParent Directory -admin/15-Sep-2010 19:19-agent/15-Sep-2010 19:21-common/15-Sep-2010 19:18-customer/15-Sep-2010 19:20-Apache/2.2.9 (Debian) PHP/5.2.6-1+lenny8 with Suhosin-Patch Server at Att, Flavio Roberto

Hello, is it possible to change how are caller id on incoming call from isdn, capi lines displayed od sip phones ? ( e.g. SNOM ) standard is 1234567@domain.net. I just want only 1234567 to be displayed. is it possible ? regards Marian -- SUNTEQ s. r. o. Hviezdoslavova 9 # Prievidza # 971 04 # Slovak republic Tel: +421-46-5430 754 # Fax: +421-46-5439 144 http://www.sunteq.sk/