similar to: AMI Redirect both calls from a bridge

Hi Neil, Am Mittwoch, den 17.12.2014, 09:08 -0500 schrieb Neil Cherry: > Doe anybody know of a way to redirect both channels from a bridge to > different dial plan extensions from the using the AMI. > > Currently, as soon as I redirect one of the channels the other appears > to be dropped and gets reorder tone (congestion, fast busy). > > I guess what I really need is a

Hi! I'm building an application which allows to dial via the Asterisk Manager Interface using the originate command. There should be an optional conferencing feature. The manager commands are basically: --------------------------------- action: login username: sdjklgdsjg secret: xxx events: on action: originate callerid: 3847438609 priority: 1 exten: 4068439865 async: 1 context: out

Originally posted on asterisk-dev with no response for 5 days, so posting it to the wider audience now. Asterisk Release 1.6.1.1 Scenario:- 1. 2 SIP peers (Zoiper softphone, if it matters) registered as 901 and 902 2. Using AMI, 901 is Originated 3. When 901 answers, it is Redirected to an extension "exten => dial,1,Dial(SIP/902)" 4. 902 rings, then answers 5.

Hello! I am having problems figuring out how to do something, and any help would be much appreciated. I would like to use the manager API to take an existing call on a specific SIP extension, dial and conference in a third party. From what I can tell, the way to do this would be to take the two original parties on the call and stick them in a meetme room using Redirect with ExtraChannel,

Is anyone else using the AMI with res_fax_spandsp.so for real-time status? I am working on migrating a FAX application from res_fax_digium.so to res_fax_spandsp.so. I have noticed that the spandsp module generates far fewer AMI status events than the Digium module and the generated events contain less information. For example when sending a fax there is no longer an event for every page. There

I'm afraid I'm quite confused by what I've found on the Wiki. I have the following dial plan that works: exten => 2201,1,Dial(sip/2201@gs1.uucp,20,) exten => 2201,2,Voicemail(u2201) exten => 2201,3,Hangup exten => 2201,102,voicemail(b2201) exten => 2201,104,hangup When the phone is in use it goes to voice mail as busy. When not picked up, as

I've been scratching my head trying to think of a way to do this, but without success yet. I'm using the Manager API. If I have two channels linked to each other (i.e. direct connection), or even if they are independent channels, I can transfer them both to the same extension by using Action: Redirect and using Channel: for one and ExtraChannel: for the other. This is most useful for

Hi All, I use the Asterisk Manager Interface to redirect the channels. There have two channels : SIP/voip_out_22-809c (None) Up Bridged Call(SIP/612-5456) SIP/612-5456 s@macro-monitor:10 Up Dial(SIP/0882@voip_out Then I send a redirect request like below : Action: Redirect Channel: SIP/612-5456 ExtraChannel:

I am trying to block/hide outgoing caller id on a PRI E1. It seems that it should be fairly simple, but it is defeating me. https://wiki.asterisk.org/wiki/display/AST/Function_CALLERID says: "to hide your caller id, use: Set(CALLERID(num-pres)=prohib)" That doesn't seem to do it. The calls are originated from AMI and I have tried a blank "CallerId:" line and

Hi all, I am trying to setup a three party conference using the Asterisk Manager API. I am using the Redirect action over an established two party call. The procedure I am using is to try to redirect the two existing channels to a third party. I would expect this to connect both channels to the third party. However, one of the two parties gets disconnected. Is this the expected behavior? Is there

There is finally an introductory book about Asterisk! It looks like Paul Mahler at www.signate.com wrote it with a lot of help from Digium. I looked at the sample pages, it looks great. __________________________________ Do you Yahoo!? New and Improved Yahoo! Mail - Send 10MB messages! http://promotions.yahoo.com/new_mail

OK, this may seem to be an obvious question but where do I find the reference docs? I'm getting this error message: Timeout, but no rule 't' in context 'home' about this line: exten => 2201,1,Dial(${PHONES1},20,Ttm) I know the problem is with the 't' but I don't know what the parameters mean. I looking for a man page basically. -- Linux Home Automation

I'm seeing all sorts of problems and it's probably more of my lack of experience than anything else. I have a BT100 running 1.0.6.7 code. When I go to the status page it says it's not registered (hmm, that's not good). I also can't get dial tone but I can dial! I'm afraid I'm lost any good pointers? I've done a sip debug and all I'm seeing for the BT100 -

Does the ${BLACKLIST()} function allow for values other than 1 to be returned and if so how can I use that is the AEL? Can I use the function in a switch statement? -- Linux Home Automation Neil Cherry ncherry@linuxha.com http://www.linuxha.com/ Main site http://linuxha.blogspot.com/ My HA Blog Author of: Linux Smart Homes For Dummies

Hi all, I'm wanting to take an existing call, and transfer both sides of it into a meetme room (yes I know the phones have a conference ability built-in but humor me). What seems to happen is I can transfer one half of it fine, but as soon as I do that the other half hangs up. Do I have to park it briefly? If so, what does the call ID become once it's parked, so that I can

Hi, I'm working on building a cluster on AWS atop CentOS 7. For development, I've been working in the eu-west-1 (Ireland) region, where the AWS MarketPlace provides an official CentOS 7 AMI (ami-e4ff5c93). However, the production deployment is taking place in AWS's GovCloud region for regulatory reasons, and there, I couldn't find an official CentOS 7 AMI. Are there plans to

Using Dahdi/PRI, I end up with channel names like 'DAHDI/i8/9995551212-4d6B', but when I do a 'core show channels' it cuts off those names to only 'DAHDI/i8/9995551212-'. This is the same for AMI. Is there a way to get the full channel name within AMI? I'm using asterisk 11.7.0 Thanks, -Justin -------------- next part -------------- An HTML attachment was

Hi folks, We are upgrading from AMI 1.0 to AMI 1.3 and looking for any documents or AMI 1.3 Specifications. I found AMI 1.4 Specification in wiki.asterisk.org but not for AMI 1.3. Can someone provide me the link for AMI 1.3 specification ? Thanks in advance Shishir -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello List, I was writing something in PHP that connects to AMI and sends a data stream ( example of it: http://slackware-es.com/ami-input.txt ), but the file (voicemail.conf , in this case) does not get fully written. I tried pasting the stream directly through telnet to AMI, and everything *appears* to be OK, but the file is not being completely written. No errors on CLI No errors on AMI

I am looking for a way to have an agent execute an attended transfer using the asterisk manager interface (AMI). I have been trying to use the dual Redirect from svn trunk. The problem with this function is that the "ExtraChannel" does not get redirected properly afaict. Now, I am looking for other solutions for the list, I will probably try playing DTMFs on the agent channel to