search for: dialpan

Hi to all. What can i do if a customer needs to log in the CDR all the dialpan actions related to a call? I mean, not only the lastapp e the lastdata but all the dialpan actions! I know that the actual CDR system store one record for each call (and for billing purposes this can be correct) but in some cases the approach needed is something similar to the queue_log. I know t...

Hi i have a similar dialplan: exten => 99,1,Gotoif(....?2:3) exten => 99,2,Meetme(100) exten => 99,3,Meetme(100|options) i'd like to do something like: exten => 99,1,Gotoif(....?2:4) exten => 99,2,Meetme(100) exten => 99,4, ... exit ... exten => 99,3,Meetme(100|options) How can i exit the dialplan? I won't use meetme twice! Thanks nik

...I am new in this list right now i am working on asterisk server and deploying asterisk PBX in my organization now i have alread setup Avaya PBX and i want to intergrate my asterisk through mediant 2000 [asterisk]-----[mediant 2k]--------E1-trunk------[Avaya] this is my setup now i want to create dialpan so how to forward call in to existing avaya setup means i have not good knowledge of dialpan routing call is there any configuration example to router call on asterisk this is possible to do in this setup suggest me Regards Satish Patel --------------------------------- Sick se...

Hi list, someone has successfully change different ringtone from dialpan with asterisk with this model Granstream? for example: exten => 0,1,Playback(pls-wait-connect-call) same=> n,SIPAddHeader(Alert-Info:;info=ring3) same=> n,Dial(SIP/310&SIP/318,30,t) can not get it to work any idea o tips? regardss -- rickygm http://gnuforever.homelinux.com

I am looking to issue a 'System cmd' that will echo the results into a new variable -- can anyone tell me the correct way to do this? example: at the terminal I am currently able to issue this command: echo | date > date.txt I end up with a file called date.txt with the contents: Mon Aug 29 18:32:45 EDT 2005 I would like to issue a command in my dialplan that puts the date

Hi, I have problem with Asterisk 1.6.0.1. I need to change language for playing prompts in Lithuanian. But in Asterisk 1.6.0.1 version everytime plays in English, but in Asterisk 1.4.x I haven't any problem. Maybe it is a bug ...? So I paste my test dialpan and prompt's locations. I hope this helps you. Files are: [root at voip asterisk]# find /var/lib/asterisk/sounds/test -name '*.wav' /var/lib/asterisk/sounds/test/lt/enter-conf-pin-number_8.wav /var/lib/asterisk/sounds/test/enter-conf-pin-number_8.wav Dialplan: [test-prompt] exten =&...

Hello, Is it possible to set a variable in dialpan when the someone calls a particular DID number so that i can use that variable for calls coming to that number only. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100819/25402ade/attachment.htm

...d a custom made rule. With FreePBX, the outgoing dialplan includes something like this: exten => _5XXXXXXXX,1,Macro(dialout-trunk,1,${EXTEN},,) exten => _5XXXXXXXX,n,Macro(dialout-trunk,2,${EXTEN},,) exten => _5XXXXXXXX,n,Macro(outisbusy,) ; trunk 1 is g0, trunk 2 is g1 If I use a custom dialpan that looks something like this: exten => _5XXXXXXXX,1,Dial(Zap/g0/${EXTEN}) exten => _5XXXXXXXX,n,NoOp(${DIALSTATUS}) exten => _5XXXXXXXX,n,Dial(Zap/g1/${EXTEN}) exten => _5XXXXXXXX,n,HangUp() and then watch the CLI, I get exactly the same behavior as above, ie. I don't get past D...

...serverlinux at gmail.com> Sent: Thursday, March 12, 2015 2:42 PM To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users at lists.digium.com> Subject: [asterisk-users] GXP 1405 and asterisk Hi list, someone has successfully change different ringtone from dialpan with asterisk with this model Granstream? for example: exten => 0,1,Playback(pls-wait-connect-call) same=> n,SIPAddHeader(Alert-Info:;info=ring3) same=> n,Dial(SIP/310&SIP/318,30,t) can not get it to work any idea o tips? regardss -- rickygm http://gnuforever.homelinux.com -- _...

We have some extensions in our dialplan that start with a star. We can dial them from Zap phones and SIP phones, but not from phones connected to a PAP2. After the user presses star follwed by two digits (our extensions are dialed with star followed by three digits) he hears a fast-busy that comes from the PAP2, not from Asterisk. This also happens with the builtin *8 (call pickup). In

We have a asterisk installation we are using for a hosted PBX solution.. we chose to use 10 digit extensions... We separated our customers by contexts and have encountered a problem where one customer can't call another using 7 digits.. even if we prepend the area code when 7 digits are dialed... anyone consider reviewing it and making recommendations? We're reluctant to post

...is my scenario: A is a SIP Phone registered on the SIP PBX "test" B is a SIP Phone registered on the SIP PBX "test" Asterisk is registered on the SIP PBX "test" with the user C D is a SIP Phone registered on Asterisk. 1) A dial C 2) C (that is Asterisk) execute the dialpan and dial D 3) A and D talks directly as the native bridging is enabled by canreinvite=yes and the codecs are compatible 4) D transfer the call to B What is the configuration needed for the 4th action? My aim is to make a REFER to B at test and free completely Asterisk. Thanks to all in advance, b...

...up (like a network interface) * number of outbound call sent * SIP status (peer reachable) * DAHDI "show channels" equivalent * queues status (number of calls proceeded, availability, ....) * "Custom " oid, dialplan controlled, and/or dialpan readable * Accessing internal * database could be nice too For examples i would like to graph the numbers of calls received and sent on a channel type like a network interface (for monitoring PRI/BRI usage for exemple). However the current values is a realtime snapshot, not the best for this...

...to make a loop test between digium card E1 to test the configuration of dahdi What I want to do scenario is I connect port 1 and port4 in the digium card with E1 cable SIPcall-->E1 Digium port 1--->(Loop)E1 port 2---->sip extension local. kindly can any can help me to draw this dialpan in the extensions.conf Description Alarms IRQ bpviol CRC4 Fra Codi Options LBO T4XXP (PCI) Card 0 Span 1 OK 0 0 0 CCS HDB3 CRC4 0 db (CSU)/0-133 feet (DSX-1) T4XXP (PCI) Card 0 Span 2 RED 0 0...

I have an asterisk box registering 100 numbers on a voip provider. Numers are: 2546.1000 to 2546.1099 My problem is that every incoming call arrived to number 2546.1099 that is the last number to register on voip provider. The correct is call arrive in destination number. See this exaple: I call to 2546.1000. -- Executing Dial("SIP/25461099-08738060", "Zap/g1/3000") in new

Hi List, I have a scenario that I want to try out (we potential have a client who would need this), but I am as of yet unable to find much help with it. What we want to do is have an asterisk box with a large number of extensions (1000+). This asterisk box will have approximately 3 SIP trunks setup back to providers. What we want to do is to be able to define groups of extensions that use