similar to: update CDR fields after Queue

Dear Mr,Ms; I am planing for a custom IVR, for example to act as a simple installer! I mean there is some choice via 0-9 and # as *Next* and * as *Back* button. is there any way for me to detect if the caller pressed # vs * on Dialplan ? -- Regards, Ali R. Taleghani 0936 322 4069 -------------- next part -------------- An HTML attachment was scrubbed... URL:

Dear Mr/Ms; web have some Queues and our Call Center and put caller in Queue Based on some regional decisions. by the way, after the Caller placed on Queues, we like to be able to reorder them on our rules. as an example: there is a queue which have 10 caller in waiting stage right now, one with the no:7 is VIP! so we need to change her place to no:2. ** again: i don't need to just make

Hello I can confirm that the variable DIALEDPEERNAME contains the information that I would expect in the variable BRIDGEPEER. But I read nowhere that DIALEDPEERNAME has replaced BRIDGEPEER as of Asterisk version 13 ?! So if this is not the intention, then yes this is probably a bug and should be reported. Kind regards. Jonas. On 18-09-16 19:58, Ludovic Gasc wrote: > Hi, > >

Hello a call goes out and is answered : [Sep 17 11:29:52] VERBOSE[23420][C-00000051] app_dial.c: SIP/myprovider-0000010b is making progress passing it to SIP/mysippeer-00000108 [Sep 17 11:30:05] VERBOSE[23420][C-00000051] app_dial.c: SIP/myprovider-0000010b answered SIP/mysippeer-00000108 [Sep 17 11:30:05] VERBOSE[23522][C-00000051] bridge_channel.c: Channel SIP/myprovider-0000010b joined

hello all, i'm really trying to mount a PDC with samba-2.2.8a and RedHat8. when i introduce the clients to domain all run Ok, but when i reboot the system and try to logon to the domain, the system send me a message like this "the system could not log you on because the domain 'DomainName' is not available" any body know what is the reason for? please help me... Note:

I'm wanting to capture the zap channel that a sip channel has connected to. I came across the ${BRIDGEPEER} variable documented on the wiki, and if I show channel SIP/<channel> when a call is connected I can see BRIDGEPEER as one of the channel variables. However ${BRIDGEPEER} is not set when I want it: I run a macro when the call has been connected. Does anyone have a hint on how

Hi, i am using the n-way-call dialplan solution found on voip-info. i have added its entry in applicationmap of features.conf file. the problem is......its not working. to activate the n-way call i dial *0 but nothing happens. i have played around with dtmf and codec settings but no success. the extensions and sip configuration is below if you want to have a look. I dont have any clue why its not

Trying to set some CDR fields in the "connected" macro of a queue command. None of the custom fields I set are stored in the database, but I can set "userfield" and it does get set. I think that the macro runs on the agent's channel, not the caller's, and this might contribute to the problem. From the sample below "userfield" (and its alias operatorid)

thanks for your reply but i think ${BRIDGEPEER} will work only when both channels are connected. i want to get channel-id before dialing so that i can dial using that channel id. > ${BRIDGEPEER} is probably a good way to do what you want.. if Channel > A calls Channel B, and you want Channel A to "get" the channelID of > Channel B, as long as the two channels are bridged,

Hi, I think this have been talked over several times but I couldn't find any answer. Sorry for asking. I want from dialplan, to transfer a callee to a context-extension-priority that would play a given fax file to callee (callee is supposed to be a fax number). I can get caller's channel id (with built-in CHANNEL variable). I found BRIDGEPEER but its value remains unset (see bellow)

Hello, SayNumber() app embeds a smart logic to play numbers according specified locale. For example, SayNumber(71) would play files "sixty", "and" and "eleven" in french. If I'm not mistaken, if I had to set a dialplan variable with the file list SayNumber would play for a given number, I would have to somehow duplicate SayNumber() logic. What would you about

Hi, I am trying to achieve 3-way conferencing taking hint from wiki link http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO Here is the scenario: 1. user "user1" calls user "user2" 2. "user1" then presses the feature code "*0" to redirect "user2" to conference room 300 3. "user1" then dials the user "user3" 4.

Hi, Currently, it seems impossible to use the output of SayNumber application as an input to Read application. So, if you want to develop an IVR with something like "You've got 23 messages. Type 1 to listen to the first one. Type 2 to leave", you must parse this message into 3 pieces and want for the last one play to start listening of user input : Playback ( "You've

cdr_mysql doesn't set the userfield when it's set inside a macro called from a feature (1.4.25, addons 1.4.8). I have a feature code: autorecord => *1,self,Macro,apprecord The apprecord macro looks like: [macro-apprecord] exten => s,1,Playback(beep) exten =>

I have A@H 2.1, running * 1.2.1. I am trying to put information into the userfield with SetCDRUserField and AppendCDRUserField. However, the field is never populated in the cdr - I've checked the csv files and the MySQL asteriskcdrdb table. The field is defined in the MySQL table, but is always empty. The csv files that get created don't have a userfield at all, that is, there

I'm tearing (what's left of) my hair out on this one :-( <shortform> How can I set the CDR(userfield) in the calling thread from the dialplan (actually a macro called from a feature) in the called thread? <long version> I use mixmonitor to record calls driven by entries in the asterisk database for selected phones. As part of this dialplan, I set the CDR(userfield) to the

Paul, Thanks - it worked! For the record, this is exactly what I did: cd/usr/src/asterisk/cdr grep -in "userfield" cdr_csv.c (to find the line that had "#define CSV_LOGUSERFIELD 1" commented out) Opened cdr_csv.c and removed the /* and */ comment marks Saved & exited After shutting down * I went to /usr/src/asterisk and did the usual: Make clean Make Make install I also

Hello, I'm using Asterisk with Realtime extensions and ODBC CDR. And I'm have some trouble with the CDR userfield that is not changed when using the SET command in the realtime dialplan. In my dialplan (extensions.conf, the file) I'm setting the userfield like this : exten => s,n,Set(CDR(userfield)="X") Later, my dialplan switches to the realtime part and this is an

Back to basics here. I want to match on one OR two digits. The following two both work, but ONLY for more than one digit, which is not as expected from the docs (see below). exten => _X.,1,SayNumber(${EXTEN}) exten => _[0-9].,1,SayNumber(${EXTEN}) This next one will ONLY match 2 digits, as expected, but the first two SHOULD match one or more, right? exten => _XX,1,SayNumber(${EXTEN})

Dears, I need to save some information on userfield when calls end in Asterisk13, but I have two error cases: 1. With endbeforehexten=no in cdr.conf, I have a registry in cdr, but userfield is not set. 2. With endbeforehexten=yes, I have two lines in cdr, one with duration, src e dst correct, and a second line with userfield setting and dst h. I am using cdr_odbc.conf, with Asterisk11.14.0 it