similar to: Dialplan - working out when users answer

Hello Guys, I am trying to convert files that are .wac to mp3 after mixmonitor command is called but it doesnt execute the command, I tried the command in terminal it worked, any help please ... below is my dial plan exten=6500,n,Set(MIXMONITOR_EXEC=&& nice -n 19 /usr/local/bin/lame -b 8 -t -F -m m --bitwidth 8 --quiet "/var/spool/asterisk/monitor/${CALLFILENAME}.wav"

I have this code in a dial plan: exten => _417XX,n,GotoIf($["${CALLERID(num)}" > "SIP/41799"]?notfromlocal) exten => _417XX,n,GotoIf($["${CALLERID(num)}" < "SIP/41700"]?notfromlocal) The value of "${CALLERID(num)}" appears to be "SIP/41712-00000181" -- Executing [41720 at from-internal:5]

Hi, I have small confusion in my configuration which is I had some DID's like 044578900-04457999. I was configured dial plan below mention. exten => _0XXXXXXXXX,1,NoOp(Int exten:${CALLERID(num)}) exten => _0XXXXXXXXX,2,Set(outgoing_ident=0445789${CALLERID(num):-2}) exten => _0XXXXXXXXX,3,NoOp(Ext ident:${outgoing_ident}) exten =>

While playing with DB function in Dialplan, I have added some garbage in AstDB. These are some family names with space in them. See this, demo*CLI> database show /18-05-2011 00:00:0052011175221575/TESTDATE : 2011-05-14 21:33:46 /18-05-2011 00:00:0052011175221575/TEST1 : 410 /18-05-2011 00:00:0052011175221575/TEST2 : 155 /18-05-2011 00:00:0052011182614252/TEST3 :

Hi All, I have been working on a project where I need to record a call in Asterisk and then split the recording into multiple audio files based on a presence of particular sound (i.e. beep) in a recording. I know this is out of scope for Asterisk but I wanted to benefit from someone else's experience if it has been done earlier. I have googled a bit and seems that Audio fingerprint(

Is there any option in Asterisk distribution of this? Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120531/8fac6a22/attachment.htm>

Hi All, I have Asterisk 11 talking to Avaya over SIP trunk using TLS and SRTP. On incoming calls from Avaya asterisk complains of 'unsupported crypto parameters: UNENCRYPTED_SRTCP' and rejects the call with '488 Not acceptable here' Doesn't Asterisk support UNENCRYPTED_SRTCP as crypto parameters in sdp? FYI SDP looks like this. v=0 o=- 1429194215 1 IN IP4 XX.XX.XX.XX s=-

I am stumped In features.conf,I programmed this [applicationmap] Answer0 => 0,self/both,Macro,nway_start But do I pass an argument or parameter to my macro? I tried Answer0 => 0,self/both,Macro,nway_start^0 Answer0 => 0,self/both,Macro,nway_start,0 but the usuar variable ${ARG1} is empty in my dialplan. The issue is that my macro needs to know what key was pressed. -------------- next

HI folks, I have set a barging facility with our production box.Client able to barge a agent but client raise a requirement, they want talk to barge agent but that communication is not listen by customer. It is possible with asterisk or not. thanks in advance. Regards Akhilesh -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, Will Keypress option will work when am in the queue and hearing MoH? Lets say a caller is waiting in queue and while he is hearing MoH, can he key in some DTMF and go to some other queue? is that possible? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, I 'm looking for a way to pass the '302 moved temporarily' received from the SIP device back to the SIP provider. Here is the setup: Some SIP phones are connected to an Asterisk System version 1.8. External connection to the public network is also done via SIP to a VoIP provider. Phone A has a CFW all calls to a phone number in public network (Mobile Phone) incoming call to

Trying to figure out the best way to pull an active call out of a queue by unique id and put it on hold. I don't want to put it on hold on the agent's phone but I want it to be pulled away from the agent's phone and into Asterisk limbo somewhere. Shortly after I want to pull the same call out of limbo and redirect it back to either the same agent or another. I was thinking about call

Hi all, I have to questions about queues. Member is a phone like SIP/myphone and only one member in the queue. At first, DIALSTATUS doesn't return any status. How to now if a call in queue has been answered or if caller just hangup? Second, how to deal with timeout, I have strange behaviors. If I put timeout=60 in queue.conf and I call the queue passing also 60 as timeout value,

hello all, i have two asterisk server for call transfer and one more asterisk server for agent login(server_X) where agent take the call. server_A and server_B server_A is connected with pri and configure with 60 channel for call transfer into server_X server_B is connected with pri and configure with 30 channel for call transfer into server_X my query is that some time two call originate same

Hi, Can a call to multiple devices be cancelled in all of them at same time? With next dialplan, exten => 100,1,Dial(SIP/101&SIP/102) when a call rings on 101 and 102 and one of them rejects the call "with 486 Busy here", is it possible to reject the call in the other device at same time? I read application dial options but I can't find any that can help me to achieve this

Hi, I use asterisk 1.8. My issue is : I have the same SIP members added to two queues. I use realtime configuration and has set the field ringinuse=0 for both the queues. But if an extension is answering the call in one queue, and some new call comes in the second queue, still that extension is ringed. In the queue_log table I am getting RINGNOANSWER events each second for the extension until

Hello everybody i am Ashish here. i am new to this mailing list. so dont know rules and regulation, just trying to post my problem of voicemail.conf Actuallt right now i am using Asterisk 1.2 on my LAN environment. i am able to call all my extension very nicely. Right now i am trying to deploying voicemail facility for all extensions, so if anybody is not present, then he/she can leave message,

Priorityjumping is totally ignored by my asterisk (tested 1.4.4 and 1.4.7.1 on FreeBSD 6.2) [general] priorityjumping=yes With n+101: exten => 1337,1,Dial(SIP/zytek,5,Ttj) exten => 1337,102,Dial(SIP/zytek,${RINGTIME},${OPTIONS}) exten => 1337,n,Hangup -- Executing [1337 at firma:1] Dial("SIP/113-087a3000", "SIP/zytek|5|Ttj") in new stack -- Called zytek

We have now got our new PRI line (10 channels, 100 numbers) connected and everything is working except the outgoing caller ID. Whatever SIP phone I'm using, the CID that's shown is the very first number... ----- s n i p ----- [default] include => outgoing include => priin [outgoing] exten => _NXXXXX.,1,Macro(dial,08${EXTEN},${RINGTIME}) ; Local number (w/o areacode) -

Hey Guys, I knew this is stupid question but i just want to know what is the difference between Queue member logged out vs Pause ? -Satish -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110321/7646a66b/attachment.htm>