search for: channelredirect

Hello, I am implementing a small ACD system on Asterisk 1.6.2.17.2 I need help with ChannelRedirect. I have a caller connected to an agent. The agent may request additional help by consulting another department. I can't use manual process with blind or directed transfer as the agent have many different numbers to dial. The message with the proper dial number is coming from the host. I got tha...

Hi, i'm trying to implement the following scenario: - A user calls number 700 - Asterisk then dials to extensions 100, 200, 300, 400 and 500 - And then bridges all calls to a conference room I tried to use MeetMe and ChannelRedirect, but seems that after channel redirect nothing more is executed. So, this seem to work for the caller and first called, but the others stay outside. Could anyone help or give me a hint? thanks Rafael

...sterisk. What I want is the possibility for the called part to push a number sequence (for example *#) to redirect the callee to a fixed extension or (for example *123#) to redirect the callee to extension 123. I'm under the impression this must be done with dynamic feature (applicationmap) and ChannelRedirect() because of the SCCP protocol. The result is that my test code, which is far from finished, only works occasionally. What happens when it fails is that the callee is hung up instead of put into the dynamic conference. There doesn't seem to be a reason for it to fail when it fails. I have tr...

...Asterisk CLI == Starting SIP/180-108c94b0 at call-third-party,s,2 failed so falling back to exten 's' == Starting SIP/180-108c94b0 at call-third-party,s,2 still failed so falling back to context 'default' Do you have some idea to achieve this kind of result ? Maybe I can use ChannelRedirect from Asterisk 1.4 ? Cna you give me a hint on that ? Tnx in advance Regards Edoardo Serra WeBRainstorm S.r.l. -- Ing. Edoardo Serra WeBRainstorm S.r.l. Via Pio Fo? 83/C 10126 - Torino Tel: +39 011 678 100 Fax: +39 011 678 275

...n a x86_64 running Linux on 2013-01-18 19:52:25 UTC How can I set variable in one context and then Redirect a channel to another context and use variable there? The code below doesn't work, so I've got empty VAR1 in context_2 [context_1] exten => s,1,SET(__VAR1=VALUE1) exten => s,n,ChannelRedirect(${CHANNEL},context_2,AMD,1) [context_2] exten => AMD,1,NoOp(VAR1: ${VAR1}) Thank you in advance, Igor -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140220/fa3c9305/attachment.html>

...T}@dynamic-nway-dest,,g) exten => 0,n,Set(DYNAMIC_FEATURES=) exten => 0,n,Goto(dynamic-nway,${CONFNO},1) exten => i,1,Goto(dynamic-nway,${CONFNO},1) [dynamic-nway-dest] exten => _XXX,1,Dial(SIP/${EXTEN}) [macro-nway-start] exten => s,1,Set(CONFNO=${FindFreeConf()}) ;exten => s,n,ChannelRedirect(${BRIDGEPEER},dynamic-nway,${CONFNO},1) exten => s,n,ChannelRedirect(${BRIDGEPEER},dynamic-nway,${CONFNO},1) exten => s,n,Read(DEST,dial,,i) exten => s,n,Set(DYNAMIC_FEATURES=nway-inv#nway-noinv) exten => s,n,Dial(Local/${DEST}@dynamic-nway-dest,,g) exten => s,n,Set(DYNAMIC_FEATURES=...

...Dial(Local/${DEST}@nway-conf-dest,,g) exten => 0,n,Set(DYNAMIC_FEATURES=) exten => 0,n,Goto(nway-conf,${CONFNO},1) exten => i,1,Goto(nway-conf,${CONFNO},1) [nway-conf-dest] exten => _.,1,Dial(SIP/${EXTEN}) [macro-nway-conf-start] exten => s,1,Set(CONFNO=300) exten => s,n,ChannelRedirect(${BRIDGEPEER},nway-conf,${CONFNO},1) exten => s,n,Read(DEST,dial,,i) exten => s,n,Set(DYNAMIC_FEATURES=nway-conf-inv#nway-conf-noinv) exten => s,n,Dial(Local/${DEST}@nway-conf-dest,,g) exten => s,n,Set(DYNAMIC_FEATURES=) exten => s,n,Goto(nway-conf,${CONFNO},1) [macro-nway-con...

...I'm working on some dialplan rules to pull multiple users into a conference call. I have some fairly straightforward rules which start up a new MeetMe conference, allow escape with the * key to invite more users, then use a features.conf sequence to bring the new user into the conference with ChannelRedirect. The problem I'm running into is the time in the MeetMe conference doesn't seem to show up in the CDRs anywhere. I tried creating the MeetMe conference, bringing one user in, bring another in, then keeping the conference open for 4 more minutes. I ended up with 4 CDR entries. None of th...

Hi, I am using asterisk via AGI and want to be able to record a call. The scenario is: 1. A call comes in 2. The call is redirected to a mobile number via a local extension and ChannelRedirect 3. The local extension looks like something this: exten => _X.,1,Dial(SIP/${EXTEN},60,?) exten => _X.,n,Agi(agi://localhost/aj.agi?action=??..) I have looked through all arguments of Dial but haven't found any way to continue having a connected call between the caller and the callee...

...dialed 1500 extension. In the extensions.conf i set as follows. exten=>1500,1,AGI(localhost// hello.agi. This hello.agi when connected plays a greeting message. Once this is connected from the script i want to transfer the call to another extension say 1600. How do i achieve this. I tried using ChannelRedirect but it didnt work. I want this transfer to happen from the script. Any suggestions please? Thanks in Advance, Jahnavi. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100728/3e7ee6fa/attachment.htm

Hi How to set a threeway conference in asterisk only for VOIP (I am using only SIP channel). Thanks Nikhil

...0,n,Dial(local/${DEST}@nway-conf-dest,,g) exten =>0,n,Set(DYNAMIC_FEATURES=) exten =>0,n,Goto(nway-conf,${CONFNO},1) exten =>i,1,Goto(nway-conf,${CONFNO},1) [nway-conf-dest] exten =>_XXX,1,Dial(SIP/${EXTEN}) [macro-nway-conf-start] exten =>s,1,Set(CONFNO=300) exten =>s,n,ChannelRedirect(${BRIDGEPEER},nway-conf,${CONFNO},1) exten =>s,n,Read(DEST,dial,,i) exten =>s,n,Set(DYNAMIC_FEATURES=nway-conf-inv#nway-conf-noinv) exten =>s,n,Dial(local/${DEST}@nway-conf-dest,,g) exten =>s,n,Set(DYNAMIC_FEATURES=) exten =>s,n,Goto(nway-conf,${CONFNO},1) [macro-nway-conf-ok...

...e(1000) exten => s,n,Return it seems to be called and seem to enter into to call but immediately hangup. first of all, in this case when in the hangup handler I will NOT like to hangup how should I finish the marco?: Hangup Return MacroExit how to redirect the call to the queue?: Queue(1000) ChannelRedirect(${CHANNEL},,1000,1) Gosub(ext-intercom,*801000,1()) dial-one,HhTtrM(auto-blkvm),1000 and what is the reason I can’t put the call back to the queue? I know that I'm already in the hangup sequence, but still wouldn't like to hangup. or this can't be done in the hangup handler? thank you...

hi all, I have a little question about meetme in Asterisk. One of my client ask me that all call can, if is necessary, become conference for 3-4 user during conversation. I think that are 2 way for make this: 1- all call (instead if the users are only 2) are conference 2- using n-way call (http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO) I decide to implement the first way because

I'd like to have an extension "join" a call. That is, C can join A and B, just as if it were an analog extension phone. ChanSpy works, sort of. The problem is that once A or B hangs up, the channel is gone. With an analog extension, C would remain connected with B if A hung up. Can I throw A and B into a confbridge and then add C? Create a new channel that grabs the A

...s: - 12020, a CLI formatting improvement - 11964, added the ability to get the original called number on SS7 calls - 11873, Added core API changes to handle T.38 origination and termination (The version of app_fax in Asterisk-addons now supports this.) - 11553, Added a status variable to the ChannelRedirect() application The changes in this release include fixes for the following issues (trivial and minor issues not included): - 11960, a crash in chan_sip - 12021, a crash related to invalid formats being specified for voicemail - 11779, fix enabling echo cancellation for incoming SS7 calls - 1174...

...s: - 12020, a CLI formatting improvement - 11964, added the ability to get the original called number on SS7 calls - 11873, Added core API changes to handle T.38 origination and termination (The version of app_fax in Asterisk-addons now supports this.) - 11553, Added a status variable to the ChannelRedirect() application The changes in this release include fixes for the following issues (trivial and minor issues not included): - 11960, a crash in chan_sip - 12021, a crash related to invalid formats being specified for voicemail - 11779, fix enabling echo cancellation for incoming SS7 calls - 1174...

...n the wiki (http://www.voip-info.org/wiki/view/Asterisk+cmd+Transfer) it is reported as a partial implementation of the REFER functionality. I've tried both atxfer and blindxfer in features.conf but it seems that asterisk make a simple Dial between the two peers. I've also taked a look at ChannelRedirect(channel|[[context|]extension|]priority) but it doesn't seem to be what i need. This 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...

Hello all, My folks would like to play a message to answering machines automatically after hanging up the phone. So, when the caller dials the number of the callee, hears an answering machine, they would like to enter a code on the phone and hang up. After the hangup the message plays to the callee and disconnects. The message that is played uses text to speech that is tailored to the callee,

Dear List, My Dial command: exten => _X.,n,Dial(SIP/PBX2/1234,60,G(connect-jack^${EXTEN}^1)) exten => h,1,.... [connect-jack] exten => _X.,1,NoOp(${CHANNEL}) ; Leg A exten => _X.,2,NoOp(${CHANNEL}) ; Leg B The problem is: after answering, [connect-jack] both priorities are executed, and right after executing them call drops. Log: -- Executing [123456 at NPDB2:76]